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SpectraLink 8020/8030 Wireless Telephone
Administration Guide
Session Initiation Protocol (SIP)
January 2010 Edition
1725-36038-001
Version H
SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Patent Information
Notice
The accompanying product is protected by one
or more US and foreign patents and/or pending
patent applications held by Polycom, Inc.
Polycom, Inc. has prepared this document for
use by Polycom personnel and customers. The
drawings and specifications contained herein
are the property of Polycom and shall be neither
reproduced in whole or in part without the prior
written approval of Polycom, nor be implied to
grant any license to make, use, or sell equipment
manufactured in accordance herewith.
Copyright Notice
© 2007 to 2010, Polycom, Inc. All rights
reserved. POLYCOM®, the Polycom "Triangles"
logo and the names and marks associated with
Polycom's products are trademarks and/or
service marks of Polycom, Inc. and are
registered and/or common law marks in the
United States and various other countries. All
other trademarks are property of their
respective owners. No portion hereof may be
reproduced or transmitted in any form or by
any means, for any purpose other than the
recipient's personal use, without the express
written permission of Polycom.
All rights reserved under the International and
pan-American copyright Conventions.
No part of this manual, or the software
described herein, may be reproduced or
transmitted in any form or by any means, or
translated into another language or format, in
whole or in part, without the express written
permission of Polycom, Inc.
Do not remove (or allow any third party to
remove) any product identification, copyright or
other notices.
Every effort has been made to ensure that the
information in this document is accurate.
Polycom, Inc. is not responsible for printing or
clerical errors. Information in this document is
subject to change without notice and does not
represent a commitment on the part of Polycom,
Inc.
2
Polycom reserves the right to make changes in
specifications and other information contained
in this document without prior notice, and the
reader should in all cases consult Polycom to
determine whether any such changes have been
made.
NO REPRESENTATION OR OTHER
AFFIRMATION OF FACT CONTAINED IN
THIS DOCUMENT INCLUDING BUT NOT
LIMITED TO STATEMENTS REGARDING
CAPACITY, RESPONSE-TIME
PERFORMANCE, SUITABILITY FOR USE, OR
PERFORMANCE OF PRODUCTS DESCRIBED
HEREIN SHALL BE DEEMED TO BE A
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OF POLYCOM WHATSOEVER.
Contact Information
Please contact your Polycom Authorized
Reseller for assistance.
Polycom, Inc.
4750 Willow Road,
Pleasanton, CA 94588
http://www.polycom.com
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About this Guide
This document explains how to configure and maintain the
SpectraLink 8020/8030 Wireless Telephones using Session Initiation
Protocol (SIP).
Polycom Model Numbers
This document covers the following registered model numbers:
802X, 803X
Related Documents
SpectraLink 8020/8030 Wireless Telephone and Accessories User Guide for
SIP (1725-36165-001)
SpectraLink 8000 SVP Server: Administration Guide for SIP
(1725-36033-001)
SpectraLink 8020/8030 Wireless Telephone: SIP User Agent: Features and
Standards (1725-36037-001)
SpectraLink 8020/8030 Wireless Telephones Handset Administration Tool
(7125-36039-001)
Available at
http://www.polycom.com/usa/en/support/voice/wi-fi/wi-fi.html
Polycom VIEW Certified Products Guide (1725-36040-001)
VIEW Configuration Guide 1725-36xxx-001 where xxx indicates a
number corresponding to the type of access point)
Available at
http://www.polycom.com/support/voice/wi-fi/view_certified.html
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3
SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Deploying Enterprise-Grade Wi-Fi Telephony
Best Practices Guide for Deploying SpectraLink 8020/8030 Wireless
Telephones
Available at
http://www.polycom.com/products/voice/wireless_solutions/wifi
_communications/handsets/spectralink_8020_wireless.html
Open Application Interface (OAI) Specification (1725-36196-001)
Available at
www.polycom.com/forms/spectralink_oai_sw_dl.html
Asterisk cmd VoiceMailMain (Shows how to enter the Asterisk
voicemail system, its menu structure, and related information.)
Available at
http://www.voipinfo.org/wiki/index.php?page=Asterisk+cmd+VoiceMailMain
Customer Support
Polycom wants you to have a successful installation. If you have
questions, please contact our Customer Support Hotline at (800) 7755330.
The hotline is open Monday through Friday, 6 a.m. to 6 p.m.
Mountain time.
For Technical Support: [email protected]
For Knowledge Base:
http://www.polycom.com/usa/en/support/voice/voice.html
For Return Material Authorization: [email protected]
Icons and Conventions
This manual uses the following icons and conventions.
Caution! Follow these instructions carefully to avoid danger.
Note these instructions carefully.
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Label
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This typeface indicates a key, label, or button on SpectraLink
hardware.
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Contents
0 About this Guide ......................................................................3
Polycom Model Numbers..................................................................3
Related Documents.............................................................................3
Customer Support...............................................................................4
Icons and Conventions.......................................................................4
1 SpectraLink 8020/8030 Wireless Telephone Overview ...........11
WLAN Quality of Service ................................................................11
WLAN Security .................................................................................12
System Diagram ................................................................................14
System Components .........................................................................15
SpectraLink 8020/8030 Wireless Telephone Specifications........18
Notes on Battery Packs.....................................................................20
Startup Sequence...............................................................................21
Handset Modes .................................................................................22
2 SpectraLink 8020/8030 Wireless Telephone Configuration ......25
The Admin Menu..............................................................................25
Admin menu................................................................................28
Phone Config ...............................................................................32
Network Config...........................................................................36
Diagnostics...................................................................................43
Restore Defaults ..........................................................................44
Demos ...........................................................................................44
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Admin Menu Default Table.............................................................45
User-Defined Preferences ................................................................47
WPA2 Enterprise PEAP Certificate Enrollment and EAPFAST Manual PAC Provisioning..............................................50
3 Software License and Protocol Management ............................55
Minimum System Requirements ....................................................55
Minimum Configuration Process ...................................................57
4 SIP Integration Factors ............................................................59
Sample DHCP Server Configuration File................................60
5 Programming the SIP Handset Features ...................................63
SIP TFTP Server Configuration Files..............................................63
Proxy server commands.............................................................65
Sample SIP Configuration Files ................................................70
6 Using the SIP Handset.............................................................85
The Handset Display ........................................................................85
Menus ...........................................................................................89
Handset Operation ...........................................................................91
7 Testing a Handset .................................................................101
8 Diagnostic Tools ...................................................................103
Run Site Survey ...............................................................................103
Diagnostics Enabled .......................................................................107
Syslog Mode.....................................................................................112
9 Certifying the Handsets.........................................................117
Conducting a Site Survey...............................................................117
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Contents
10 Software Maintenance ..........................................................121
Upgrading Handsets ................................................................121
Normal Download Messages ..................................................122
Download Failure or Recovery Messages .............................122
11 Troubleshooting....................................................................123
Access Point Problems ...................................................................123
Handset Status Messages...............................................................125
12 Appendix A: Regulatory Domains .........................................135
13 Index ...................................................................................137
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1
SpectraLink 8020/8030
Wireless Telephone Overview
The SpectraLink 8020/8030 Wireless Telephones is a Wi-Fi handset
for workplace telephone systems. The handsets operate over a VIEW
Certified 802.11a/b/g/n wireless LAN (WLAN) providing users a
wireless extension of the SIP call server. By seamlessly integrating
into a SIP environment, wireless telephone users are provided with
high-quality mobile voice communications throughout the workplace,
giving users the freedom to roam throughout the workplace while
providing all the features and functionality of a wired SIP desk
phone.
In the SIP environment, the handsets support up to three SIP call
servers. Each handset may have five line appearances and two calls
per line. Each handset may have up to 10 sets of credentials to
identify itself by current user.
The latest wireless telephone and Handset Administration Tool
software versions are required to support the features described in
this document. See Chapter 3 Software License and Protocol
Management.
WLAN Quality of Service
WLAN Quality of Service (QoS) is provided by using one of three
available mechanisms: SpectraLink Voice Priority (SVP), Wi-Fi Standard
QoS, or Cisco Compatible Extensions (CCX) version 4. These QoS
modes can not be mixed within the same WLAN; therefore, all Wireless
Telephones on the network must have the same QoS setting.
SVP
SpectraLink Voice Priority (SVP) is a proprietary method of WLAN
QoS, developed by Polycom, to ensure enterprise-grade voice quality,
battery life and call capacity for SpectraLink Wireless Telephones.
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
SVP requires the use of the SVP Server, which is an Ethernet LAN
device that works with in conjunction with Wi-Fi APs to ensure QoS
over the WLAN. Voice packets to and from the Wireless Telephones
are tunneled through the SVP Server to ensure voice prioritization as
they are routed between the handset and SIP call server. See the
SpectraLink SVP Server: Administration Guide within IP Environments
document for detailed information about this device.
Wi-Fi Standard QoS
SpectraLink Wireless Telephones support WMM, WMM Power Save
and WMM Admission Control - all QoS standards from the Wi-Fi
Alliance based on IEEE 802.11e. The combination of these three
standards provides enterprise-class QoS in terms of voice quality,
battery life and call capacity. The WLAN must also support and
enable each of these QoS mechanisms in order to ensure they are
utilized. This option does not require the SVP Server.
CCXv4
The CCX program requires WLAN client devices operating on Cisco
APs to use a defined set of industry standards and Cisco-specific
features. The SpectraLink 8020/8030 Wireless Telephone has been
certified by Cisco as CCXv4 compliant. When the CCXv4 operating
mode is selected on the handset, it automatically initiates the required
set of Cisco-specific and industry standard QoS mechanisms. This
option does not require the SVP Server.
WLAN Security
The following security methods are supported by the handset.
WPA2 Enterprise
The handset supports WPA2 Enterprise, as defined by the Wi-Fi
Alliance. WPA2, which is based on the 802.11i standard, provides
government-grade security by implementing the Advanced
Encryption Standard (AES) encryption algorithm. The Enterprise
version of WPA2 uses 802.1X authentication, which is a port-based
network access control mechanism using dynamic encryption keys to
protect data privacy. Two 802.1X authentication methods are
supported on the Wireless Telephone, EAP-FAST and
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SpectraLink Wireless Telephone Overview
PEAPv0/MSCHAPv2. Both of these methods require a RADIUS
authentication server to be available on the network and accessible to
the phone. Additional details are provided in Section 3.1.
Normal 802.1X authentication requires the client to renegotiate its key
with the authentication server on every AP handoff, which is a timeconsuming process that negatively affects time-sensitive applications
such as voice. Fast AP handoff methods allow for the part of the key
derived from the server to be cached in the wireless network, thereby
shortening the time to renegotiate a secure handoff. The Wireless
Telephone supports two fast AP handoff techniques, Cisco Client Key
Management (CCKM) (only available on Cisco APs) or Opportunistic
Key Caching (OKC). One of these methods must be configured for
support on the WLAN to ensure proper performance of the handset.
WPA and WPA2 Personal
The handset supports WPA and WPA2 Personal, as defined by the
Wi-Fi Alliance. WPA2, which is based on the 802.11i standard,
provides government-grade security by implementing the Advanced
Encryption Standard (AES) encryption algorithm. WPA, which is
based on a draft version of the 802.11i standard before it was ratified,
uses Temporal Key Integrity Protocol (TKIP) encryption. The
Personal version uses an authentication technique called WPA2 is
based on the 802.11i standard. Pre-Shared Key (PSK) allows the use
of manually entered keys or passwords to initiate WPA security.
Cisco Fast Secure Roaming
Cisco’s Fast Secure Roaming (FSR) mechanism uses a combination of
standards-based and proprietary security components including
Cisco Client Key Management (CCKM), LEAP authentication,
Michael message integrity check (MIC) and Temporal Key Integrity
Protocol (TKIP). FSR provides strong security measures for
authentication, privacy and data integrity along with fast AP handoff
on Cisco APs.
WEP
The handset supports Wired Equivalent Privacy (WEP) with both 40bit and 128-bit encryption.
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
System Diagram
The following diagram shows the Polycom components residing on a
network with APs and wireless LAN Ethernet Switch.
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SpectraLink Wireless Telephone Overview
System Components
SpectraLink 8020/8030 Wireless Telephone
The SpectraLink 8020 Wireless Telephone is a lightweight, durable
handset specifically designed for mobile workplace use. The SpectraLink
8030 Wireless Telephone has the same features and function, but in a
more durable design with and includes push-to-talk capability.
Like a wired desk phone, the handset can receive calls directly, receive
transferred calls, transfer calls to other extensions, and make outside and
long distance calls. The Wireless Telephones can only be used onpremises within the WLAN coverage area with the local SIP call server.
SpectraLink 8000 SVP Server (required when using SVP)
As described earlier, the SVP Server is a Polycom wired LAN device
that is required when using SpectraLink Voice Priority for QoS.
Access points
Enterprise-grade Wi-Fi access points provide the connection between
the wired LAN and the wireless LAN. VIEW Certified 802.11a/b/g/n
APs must be positioned in all areas where Wireless Telephones will
be used to ensure seamless radio coverage. The number, type and
placement of access points will affect the coverage area and capacity
of the wireless system. Careful planning of the WLAN is necessary to
ensure good voice quality. See the Best Practices Guide for Deploying
SpectraLink 8020/8030 Wireless Telephones for additional guidance.
APs must be properly configured to support the corresponding QoS
and security methods selected for the handset.
Ethernet switch
One or more Ethernet switches interconnect multiple network
devices, including the SpectraLink 8000 SVP Server (if used for QoS),
the proxy server(s), wired IP phones, TFTP Server, RADIUS
authentication server (if using WPA2 Enterprise) and WLAN access
points. Enterprise Ethernet switches provide the highest performance
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
networks, which can handle combined voice and data traffic, and are
required when using the SpectraLink 8020/8030 Wireless Telephones.
Although a single Ethernet switch network is recommended, the
handsets and the SpectraLink 8000 SVP Server can operate in larger,
more complex networks, including networks with multiple Ethernet
switches, routers, VLANs and/or multiple subnets, as long as the SVP
Server and access points and handsets are on the same subnet.
However, in such networks, it is possible for the quality of service
(QoS) features of the SVP Server to be compromised, and
consequently voice quality may suffer. Any network that consists of
more than a single Ethernet switch should be thoroughly tested to
ensure any quality issues are addressed. See the Best Practices Guide
for Deploying SpectraLink 8020/8030 Wireless Telephones for additional
guidance.
SpectraLink 8020/8030 Wireless Telephones cannot roam with
uninterrupted service between subnets unless specific LAN
components are present. Certain AP/Ethernet switch combinations
establish a Layer-2 tunnel across subnets that enable the handsets to
roam. Without this capability, any call in progress will be dropped
when the user moves out of range and the handset must be power
cycled in order to resume functionality in the new subnet area.
Ensure that all your APs are attached to the same subnet for proper
operation. The handset can change subnets if DHCP is enabled and
the handset is powered off then back on when within range of APs on
the new subnet. Note that the wireless telephones cannot “roam”
across subnets, since they cannot change IP addresses while
operational.
SIP server
The SIP server is a component from a third-party vendor that
provides access to telephony services. The handsets can recognize up
to three distinct SIP servers in a single system.
The handsets can operate with SIP proxy servers such as SER (SIP
Extensible Router) or with SIP Back-to-Back User Agents (B2BUA) –
the most common form of SIP server for PBX-based systems. In this
case the IP address location services provided by the SIP server are
not available, so direct IP address dialing must be used.
The SIP proxy server connects to another device such as a PBX or
gateway and from there, other wired phones and the PSTN.
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SpectraLink Wireless Telephone Overview
TFTP (Trivial File Transfer Protocol) server
A TFTP server is required in the system to distribute software to the
handsets. It may be on a different subnet than the APs and/or
handsets.
SIP TFTP server
The SIP TFTP server is required in the system to deliver SIP
configuration parameters to the SIP handset whenever a SIP handset
is powered-up. The location of the SIP TFTP server is separately
specified in SIP handset administration parameters (See Chapter 2
SpectraLink 8020/8030 Wireless Telephone Configuration, section The
Admin Menu.) Normally, the SIP TFTP server is the same as the
network TFTP server, see above.
NTP (Network Time Protocol) Server
If WPA2 Enterprise security is used, the handset will confirm the
PEAP certificate has a valid date and time with the NTP Server on the
network, if one is available. If an NTP Server is not available, the
certificate will be assumed valid and operate accordingly.
Authentication Server (if using WPA2 Enterprise)
A RADIUS authentication server must be used to provide
username/password-based authentication using RSA certificates for
PEAPv0/MSCHAPv2 or PAC files for EAP-FAST.
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
SpectraLink 8020/8030 Wireless Telephone
Specifications
SpectraLink 8030 Wireless Telephone
18
SpectraLink 8020 Wireless Telephone
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SpectraLink Wireless Telephone Overview
Table of Specifications
Radio mode
(selectable)
(802.11b, 802.11g)
(802.11a)
2.4–2.4835 GHz
5.150–5.250 GHz
5.250–5.350 GHz
5.470–5.725 GHz
5.725–5.825 GHz
Transmission type
Direct-sequence spread spectrum (DSSS)
Transmit data rate
up to 54 Mb/s
WLAN QoS
SpectraLink Voice Priority (SVP)
Wi-Fi Standard QoS (using WMM, WMM-Power Save and
WMM-Admission Control)
CCXv4
WLAN security
WEP (Wired Equivalent Privacy)
Cisco FSR (Fast Secure Roaming)
WPA Personal
WPA2 Personal
WPA2 Enterprise:
802.1X Authentication
EAP-FAST
PEAPv0/MSCHAPv2:
PEAP certificate sizes: 512*, 1024*,
2048, 4096 bit (*recommended)
Encryption Ciphers: AES, RSA, RC4
Data Integrity: Hashed Message
Authentication Code MD5 (HMACMD5) (RFC 2403, 2104) and Secure
Hash Algorithm-1 SHA (HMAC-SHA1) (RFC2404)
Fast AP Handoff
Opportunistic Key Caching (OKC)
Cisco Client Key Management (CCKM)
FCC certification
Other certifications
Part 15.247
IP 53 certified for resistance to dust and liquid resistance
MIL 810F Proc IV 516.5 for shock resistance
Cisco Compatible Extensions (CCX) v4
Voice encoding
ADPCM (Proprietary) G.711µ-law, G.711a-law and G.729
Transmit power
Up to 100mW Transmit Power Control (formerly 802.11h),
see Appendix A for details.
Display
8020 Dimensions
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Up to five lines of text plus two icon status rows and one row
for softkey labels.
5.7" x 2.0" x 0.9"
(14.5 x 5.1 x 2.3 cm)
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
8030 Dimensions
5.4" x 2.0" x 0.9"
(13.7 x 5.1 x 2.3 cm)
8020 Weight*
3.9 oz. ( 110.6 g) with Standard Battery Pack
8030 Weight*
4.2 oz. (119.1 g) with Standard Battery Pack
Standard Battery Pack capacity
4 hours talk, 80 hours standby
Extended Battery Pack capacity
6 hours talk, 120 hours standby
Ultra-Extended Battery Pack capacity
8 hours talk, 160 hours standby
Notes on Battery Packs
20
•
Polycom offers the following battery types with increasing
capacity: Standard, Extended and Ultra-Extended.
•
Battery Pack life will vary depending on handset model and
features and system infrastructure.
•
Batteries are shipped with a partial charge. Polycom recommends
fully charging them before using them in phone operation.
•
Maximum Battery Pack performance is achieved after a few
charge/discharge cycles.
•
If multiple Battery Packs are supplied with your handset, Polycom
recommends that each be fully charged upon receipt to prolong
battery life. Battery Packs will slowly lose charge if unused. To
maintain battery potential, charge unused Battery Packs
occasionally or alternate Battery Pack use.
•
After a length of time Battery Packs will lose the ability to
maintain a charge and to perform at maximum capacity and will
need to be replaced. This is normal for all batteries.
•
Overnight charging is best done while the handset is turned off.
•
If the handset does not charge, clean Battery Pack, charger and
handset contacts with an alcohol swab.
•
When the handset is properly seated, the backlight comes on
briefly to indicate charging has begun.
•
Any battery which exhibits swelling, cracking or other
abnormality should be disposed of promptly and properly.
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SpectraLink Wireless Telephone Overview
Startup Sequence
The SpectraLink 8020/8030 Wireless Telephone goes through an
initialization sequence at startup. The line icons 1-9 display and count
down as the handset steps through this sequence. This is usually very
rapid. If there is difficulty at any step that prevents initialization from
continuing, an error message will display and the related icon(s) will
stay on. Please see the error table at the back of this document for
instructions on how to handle error messages that occur during
initialization.
Icon
The icon(s) shown in bold turns off when:
123456789
The handset has located and authenticated and associated with at least one AP, and is
proceeding to bring up higher-layer networking functions.
12345678
The handset is either configured for Static IP, or if configured for DHCP, the DHCP
discovery process has started.
1234567
If DHCP is configured, a DHCP response was received which contains a good DNS
server configuration.
123456
Note: Used for SVP QoS only and not present when using Wi-Fi Standard QoS or
CCXv4.
Indicates one of the following possibilities:
1. Static IP configuration
2. SVP Server address found in DHCP option 151 response
3. SVP Server address found via DNS lookup
12345
All networking functions are complete (notably, DHCP), and the handset is proceeding
with establishing the SRP link to the SpectraLink 8000 SVP Server.
1234
Note: Used for SVP QoS only and not present when using Wi-Fi Standard QoS or
CCXv4.
The SRP link is established; all network stack initialization is complete, proceeding with
application-specific initialization.
123
SIP application startup. Icon 3 is extinguished if a generic SIP config file is found.
12
Icon 2 is extinguished if a handset specific SIP config file is found.
(no icons)
Handset is attempting to register each of the specified line contacts.
Registering
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Icon
(no icons)
EXT.
XXXXX
The icon(s) shown in bold turns off when:
Handset has registered with at least one contact on one proxy server. Initialization is
complete. The handset is in standby mode ready to receive and place calls. The line one
contact is displayed.
During the last three steps of this process, the handset contacts the SIP
TFTP server and downloads general information about the proxy
server(s), downloads specific information pertaining to the handset,
registers with the proxy server(s), and verifies handset credentials.
Once this process is complete, the handset is ready to use.
If the username and password have not been defined in the Admin
menu, you will be prompted to enter both of these items before the
extension number can display. The user name must correspond to the
configuration file that contains user-specific information. If the file is
not found, an error message will appear and the handset will restart.
See Chapter 5 Programming the SIP Handset Features, section SIP TFTP
Server Configuration Files.
Handset Modes
Standby mode (on-hook)
In standby mode, the handset is waiting for an incoming call or for
the user to place an outgoing call. The extension number is shown on
the display and there is no dial tone. In this mode, the handset is
conserving battery power and wireless LAN bandwidth.
When an incoming call arrives, the handset rings; the handset enters
the active mode and remains so until the call is ended. The call is
answered by pressing the START key or the Answ or Spkr softkey. The
handset will ring according to user preference as specified in the
standby menus. The ringing can be silenced by pressing the END key.
If you do not wish to accept the call, some SIP call servers support the
ability to press the Rej softkey. If supported, the SIP call server will
redirect the call as configured by the system administrator.
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SpectraLink Wireless Telephone Overview
Active mode (off-hook)
The handset is in the active mode when an incoming call is answered
or when it is in communication with the SIP server without being in
an active call.
When an incoming call occurs during an active call, the handset will
play the second call ringing sound until the call is answered, the caller
hangs up, or the call transfers to voicemail. If the END key is pressed,
the first call is terminated and the handset reverts to a full ring.
The active mode utilizes the most bandwidth and battery power. To
conserve battery resources, return the handset to the standby mode
when a call is completed by pressing the END key.
Push-to-talk (PTT) mode
The SpectraLink 8030 Wireless Telephones utilize channels for
incoming and outgoing radio communication. While PTT is active, the
handset is in PTT mode. It can receive regular phone calls in this
mode. When a regular phone call is answered, the handset enters
active mode.
Configuration menu mode
When user preferences are being configured in the Config menu, the
handset is on but is not active. It cannot receive calls while in the
Config menu. If the handset is idle 15 seconds while in the Config
menu, it will return to the standby state. If a call was trying to be
established while in the Config menu, it will ring as soon as it is
returned to the standby state.
Messaging mode
If text messaging functions have been programmed, as in a nurse call
system, the handset is able to receive text messages. While these
messages are being accessed, the handset is in messaging mode.
Incoming calls will ring with the second call ringing sound.
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2
SpectraLink 8020/8030 Wireless
Telephone Configuration
Each handset may be configured for site-specific requirements by
opening the Admin menu and selecting options or entering specific
information. Any settings entered in the Admin menu must conform to
system settings. Only the handset being configured is affected by the
Admin menu settings.
The wireless telephone user may select several usability options from
the Standby menu, described below in the User-defined Preferences
section. This information is also provided in the end-user manual.
The SpectraLink Handset Administration Tool is a software utility
that enables rapid configuration of handsets by utilizing the USB port
on the Dual Charger. See the Handset Administration Tool (HAT)
document for specific instructions. Please see your service
representative or contact Polycom customer service for more
information about this time-saving tool.
When WPA2 Enterprise security is used, PAC files for EAP-FAST can
be provisioned wirelessly or by using the HAT. For PEAP, the HAT
must be used to enroll certificates. See section WPA2 Enterprise PEAP
Certification Enrollment and EAP-FAST Manual PAC Provisioning at the
end of this chapter for details.
Other settings that must be configured include, but are not limited to,
WLAN QoS, DSCP tagging, DHCP and regulatory domain
information. If these are not selected by the administrator the handset
will use the default settings.
The Admin Menu
The Admin menu contains configuration options that are stored locally
(on each handset). Each handset is independent, and if the default
settings are not desired, the Admin options must be set in each handset
requiring different settings.
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Opening the Admin menu
The handset Admin menu can be accessed in one of two ways:
1. With the handset powered off, press and hold the START key.
While holding the START key, press and release the END key. When
the Admin menu appears, release the START key.
2. Press and release the END key. Press and hold the START key.
When the Admin menu appears, release the START key.
If an admin password has been set, the display will require its entry
before opening the Admin menu. The default password is 123456. If
no password is set, the display will proceed directly into the Admin
menu.
Navigation
The navigation keys just below the softkeys are used to navigate
through and select menu options. These are referred to as Nav▲,
Nav▼, Nav◄, Nav►, and NavOK.
Toggle options
Some menu items have only two options, which operate on a toggle
basis. The current setting is shown below the menu heading on the
info line. The other available setting is highlighted in the menu list.
Press NavOK to activate the highlighted setting.
For example, when predial is disabled, the info line displays Predial
Disabled and the highlighted menu item is the Enable Predial option.
Press NavOK to enable predial. The info line will change to display
Predial Enabled.
In another example, when the info line displays Ring in Speaker, the
highlighted menu option is Ring in Headset. Press NavOK to select Ring
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SpectraLink Wireless Telephone Configuration
in Headset, The ring will now sound in the headset and the info line
will change to Ring in Headset.
Data entry and editing
An asterisk (*) next to an option on the display indicates that it is
selected. Use the Nav keys and the softkeys to navigate and select
desired options.
Enter numbers by pressing the buttons on the keypad. The blinking
underscore identifies the current cursor position. When entering
alphanumeric strings, the CAPS/caps softkey will appear and may be
pressed to toggle the case. Enter letters by repeatedly pressing the
corresponding key until the desired letter displays on the screen. Use
the CAPS softkey to change the case as needed.
To edit during entry, delete the character to the left of the cursor by
pressing the Del softkey. To replace an entry, delete it by pressing the
Clr softkey and then enter the new data. To edit an existing entry, use
Nav◄ and Nav► to move the cursor position, and then press the Del
softkey to delete the character to the left. Insert new data by pressing
the buttons on the keypad.
Alphanumeric entries:
caps
CAPS
1
1
1
2
2abc
2ABC
3
3def
3DEF
4
4ghI
4GHI
5
5jkl
5JKL
6
6mno
6MNO
7
7pqrs
7PQRS
8
8tuv
8TUV
9
9wxyz
9WXYZ
0
0
0
*
*.!$%&’()+,:;/\=@~
#
<space>
Key
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Admin menu
The following table lists the Admin menu items. The default settings
have an * prior to the option. Detailed descriptions of each item
appear below the table.
1st level
2nd level
3rd level
Phone Config
Telephony Protocol
*Type 036
PTT/Emerg. Button
Emergency Dial
4th level
5th level
Emergency #
[Enable/Disable]
Emergency
Number
PTT
[Enter Number]
[Enter Name]
PTT
[Enable/*Disable]
Allowed Channels
*Channel 1
*Channel 2
*….
*Channel 24
Name Channels
[list ]
Priority Channel
Priority Channel
On/*Off
Name Channel
Time Zone
[list]
*GMT
Daylight Savings
*DST No Adjust
DST Auto (USA)
DST Auto (AUS)
DST Auto (EURO)
Password
*Enable/Disable
[If Password is enabled]
Change Password
SIP Registration
Login
[for each option]
Reg 2
Username
Password
Reg 3
Reg 4
Clear SIP Regist.
OAI
28
*Enable OAI
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1st level
2nd level
3rd level
4th level
5th level
Disable OAI
Location Service
Enable RTLS
*Disable RTLS
Transmit Interval
1 minute
5 minutes
*10 minutes
Network Config
IP Addresses
Location Server IP
Enter IP
ELP Port
Enter Port *8552
*Use DHCP
Static IP
Phone IP
Default Gateway
Subnet Mask
TFTP Server IP
Syslog Server IP
Time Server IP
SVP Server IP
SIP TFTP Svr IP
OAI Server IP
SS ID
[enter]
WLAN Settings
3rd level
4th level
5th level
*Custom
Security
*None
WEP
6th level
7th level
Authentication
*Open System
Shared Key
WEP
[Enable/*Disable]
Key Information
Default Key
Key Length
Key 1-4
Rotation Secret
WPA2-PSK
*Passphrase
Pre-Shared Key
WPA-PSK
*Passphrase
Pre-Shared Key
Cisco FSR
Username
Password
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
3rd level
4th level
5th level
6th level
7th level
WPA2-Enterprise
Authentication
*EAP-FAST
PEAP
Fast Handoff
*CCKM
OKC
Username
Password
Delete
[Cert./PAC]
QoS
*SVP
DSCP tags
WT in call (*46)
WT standby (*26)
Other (*0)
Wi-Fi Standard
DSCP tags
Voice (*46)
Control (*26)
Other (*0)
Admission Cntrl
*Mandatory
Optional
CCX
WPA2-Enterprise
Authentication
*EAP-FAST
PEAP
Fast Handoff
*CCKM
Username
Password
QoS
Delete [Cert./PAC]
[Yes/No]
DSCP tags
Voice (*46)
Control (*26)
Other (*0)
1st level
2nd level
3rd level
Network Config
Reg. Domain
01
4th level
5th level
02
03
04
05
06
07
08
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SpectraLink Wireless Telephone Configuration
1st level
2nd level
3rd level
4th level
Æ
[802.11 Config]
a Æ
5th level
[ 802.11a]†
5.150–5.250
5.250–5.350 DFS
5.470–5.725 DFS
5.470-5.650 DFS
5.725–5.825
5.725-5.850
‡b & b/g mixed
g only
Æ
[Transmit Power]
5mW (7dBm)
10mW (10dBm)
20mW (13dBm)
*30mW (15dBm)
40mW (16dBm)
50mW (17dBm)
100mW (20dBm)
Diagnostics
Run Site Survey
Diagnostics Mode
*Disable
Enable
Syslog Mode
*Disabled
Errors
Events
Full
Error Handling Mode
Halt on Error/
*Restart on Error
Restore Defaults
Demos
* default setting
Graphics Demo
† Only those 802.11a bands that are available in the selected domain will be listed. See Appendix A for
complete information.
‡ Subbands have not been established for the b and b/g mixed or the g-only mode at this writing.
Provision is made in the software to accommodate these ranges once established. Until added, selecting
either of these two modes will immediately bring up Transmit Power options.
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Attention: Modifications of settings under WLAN Settings = “CCX” may get reflected in the
corresponding settings under WLAN Settings = “Custom” or vice versa. If the configuration is changed
from “CCX” to “Custom” or vice versa, it is recommended to double-check all settings.
Phone Config
Telephony Protocol
Telephony Protocol lets you select the VoIP protocol that your site is
licensed to download and run. The SIP protocol used for the
SpectraLink 8020/8030 Wireless Telephones requires license option
selection 36. Any other protocol will cause the handset to
malfunction.
PTT/Emerg. Button
This option appears only on the SpectraLink 8030. The Push-to-talk
button on the left side of the handset may be configured to either
standard PTT functionality or to dial the specified emergency call
number when pressed twice within two seconds. These are mutually
exclusive options. Both are disabled by default.
When using the Handset Administration Tool to configure this
option, ensure the PTT option in the PTT Admin tab under Handset type
is disabled before enabling the Emergency Dial option in the Phone
Config tab. When PTT is enabled, the Emergency Dial option will be
grayed out.
Push-to-talk [Disable/Enable] – If enabled, the PTT options will appear on
the Config menu for the end user to subscribe to allowed channels, etc.
If disabled, the PTT options will not appear on the Config menu and
the Emergency Dial option may be enabled.
PTT is disabled by default. When enabled, all 24 PTT channels are
allowed by default. To toggle the allowed status of any channel, select
Allowed Channels, scroll to the channel to be disallowed and press
NavOK. Allowed channels are displayed with an asterisk (*) in the left
column. Only those channels allowed in the Admin menu will appear
on the Config menu where they can be subscribed to by the end user.
The priority channel, labeled by default as channel 25, may be set and
will be available to all PTT handsets. When a PTT broadcast is made
on the priority channel, it will override any active PTT transmission
on all other channels.
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Emergency Dial – the Emergency Dial option allows you to enable or
disable the feature. When enabled, the handset will dial the number
programmed into the Emergency Number option when the panic button
is pressed twice within two seconds.
Caution! Emergency dial just sets up a telephone call and will be
inoperable if the wireless system or the call server fails for any
reason. Do not rely on it as your sole method of emergency
notification.
Follow your dial plan rules when entering the emergency number to
be dialed. E.g. if an outside number is to be dialed and a prefix is
required to obtain an outside line, enter the prefix as part of the
emergency number.
Once an Emergency Number has been entered, it can be modified,
but can only be cleared by restoring the handset to defaults.
Time Zone
Worldwide time zone options are available. Greenwich Mean Time
(GMT) is the default.
Daylight Savings
The handset may be adjusted for daylight savings time.
Password Enable/Disable/Change
The password option controls access to the Admin menu. It is enabled
by default with the password 123456. The Password option operates as
a toggle between Enabled and Disabled. The info line will display the
current state. Press NavOK to change the password protection state. To
modify the password requirement, the default or previously set
password must be entered to verify the change. Change Password will
appear only if the password is enabled. The password is disabled by
default. The password must be set in each handset for which
controlled access is desired.
SIP Registration
Individual handsets may be configured to correspond with the SIP
configuration information in the TFTP server. The handset is then
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
automatically identified at startup. If username and password
information is not configured in the Admin menu, then this
information will be requested at startup
In either case, the username must agree with a corresponding
configuration file. See Chapter 5 Programming the SIP Handset Features,
section SIP TFTP Server Configuration Files.
Login allows you to specify a username and password for
automatically acquiring SIP configuration information. If no
username is specified, the SIP handset will request username and
password at startup and any additional registrations specified here
are ignored.
The username should correspond to the primary (line 1) dial number
assigned to the user. The username and password should also
correspond to the authentication credentials as created by your
system administrator for your primary line registration. Usernames or
passwords can be erased by selecting the item, then pressing the Bksp
softkey and then the Save softkey.
Reg 2, Reg 3, and Reg 4 allow you to specify additional authentication
usernames and passwords that may be required by your handset for
any additional line appearances (registrations) that may appear in the
specific user’s configuration file. This information will be ignored if a
Login username is not provided.
OAI Enable/Disable
Polycom’s Open Application Interface (OAI) enables third-party
computer applications to display alphanumeric messages on the
handset display and take input from the handset keypad. Refer to the
OAI Specification (Version 2.0) documentation for information about
administering the OAI Gateway and the services it can provide.
If you have an OAI Gateway installed in your system, OAI may be
optionally enabled in each handset. You may select whether the
handset should attempt to connect to the SpectraLink 8000 OAI
Gateway by choosing either the Enable or Disable options in this menu.
If OAI is enabled, and an IP address (called the OAI Server IP) is
available to the handset (either via DHCP or Static IP configuration),
the handset will communicate with the OAI Gateway at power-on,
and periodically while it is powered-on. If you don’t have a
SpectraLink 8000 OAI Gateway installed at your site, you should
disable the OAI feature to preserve network bandwidth and battery
life.
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SpectraLink Wireless Telephone Configuration
Location Service
Location service may be used to enable or disable the Ekahau RealTime Location System (RTLS), select a transmit interval, or enter a
static IP address for the Ekahau Positioning Engine (EPE). Location
services capability is provided by the EPE 4.0 using Ekahau Location
Protocol (ELP). See Ekahau’s user documentation for more
information.
RTLS [Enable/Disable] The RTLS is disabled by default. Press NavOK to
toggle to the alternate setting. When RTLS is enabled, the handset will
display the RTLS icon in the top center of the screen.
The ring indicator icon will take precedence over the RTLS icon, i.e.
the new icon will not be visible while the handset is ringing. When
ringing has ceased and the ring indicator becomes inactive, the RTLS
icon will again appear (regardless of hook state).
Transmit interval Allows selection of 1 minute, 5 minutes, or 10 minutes for
maximum time between transmit intervals. Default transmit interval
is 10 minutes. Press NavOK to select the desired transmit interval.
To optimize battery life, the interval between sending out ELP
updates will vary based on handset state. It is expected that ELP
updates will occur at most every two to six seconds and at least
every few minutes. If improved tracking capability is desired, set the
transmit interval for a shorter time between ELP updates. Increasing
the frequency of transmissions will decrease battery life.
Location Server IP Allows the user to statically enter the IP address of
the EPE. Enter the IP address and press NavOK to save.
Ekahau clients are not expected to find the EPE automatically.
Regardless of the handset’s selection of DHCP or static IP, the EPE
IP address must be statically entered in the Ekahau admin menus or
HAT.
ELP Port Allows the user to select the port number which ELP updates
get sent to at the Location Server IP address. It must match the value
configured in the Ekahau Positioning Engine for proper functionality.
The ELP port number must be greater than zero and less than 65536.
Default is 8552. Enter the port number and press NavOK to save.
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Network Config
IP Addresses
There are two modes in which the handset can operate: DHCPenabled or Static IP. Select the mode for operation from the IP
Address menu:
* Use DHCP Will use Dynamic Host Configuration Protocol to assign
an IP Address each time the handset is turned on. If DHCP is enabled,
the handset also receives all other IP Address configurations from the
DHCP server.
Static IP Allows you to manually set a fixed IP Address. If selected, the
handset will prompt for the IP addresses for each configurable
network component. When entering addresses, enter the digits only,
including leading zeroes. No periods are required.
Regardless of the mode in which the handset is operating, the
following components are required and must be configured as part of
the SIP system:
Phone IP The IP address of the handset. This is automatically
assigned if DHCP is used. If using Static IP configuration, you
must obtain a unique IP address for each handset from your
network administrator.
Default Gateway and Subnet Mask Used to identify subnets, when
using a complex network, which includes routers. Both of these
must be configured either with an IP address under Static IP (not
set to 000.000.000.000 or 255.255.255.255) or with DHCP for the
handset to contact any network components on a different subnet.
If configured on the DHCP server, use option 3 for the Default
Gateway and option 1 for the Subnet Mask. Contact the network
administrator for the proper settings for the network.
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SpectraLink 8020/8030 Wireless Telephones cannot roam with
uninterrupted service between subnets unless specific LAN
components are present. Certain AP/Ethernet switch combinations
establish a Layer-2 tunnel across subnets that enable the handsets
to roam. Without this capability, any call in progress will be dropped
when the user moves out of range and the handset must be power
cycled in order to resume functionality in the new subnet area.
Ensure that all your APs are attached to the same subnet for proper
operation. The handset can change subnets if DHCP is enabled and
the handset is powered off then back on when within range of APs
on the new subnet. Note that the wireless telephones cannot “roam”
across subnets, since they cannot change IP addresses while
operational.
Please see Best Practices for Deploying Enterprise-Grade Wi-Fi
Telephony for detailed configuration information.
TFTP Server IP The IP address of a TFTP server on your network,
which holds software images for updating the handsets and
contains the handset files. If this feature is configured (not set to
0.0.0.0 or 255.255.255.255) with either Static IP configuration or
using DHCP option 66 (TFTP server), or the boot server/next
server (siaddr) field, the handset will check for newer software
each time it is powered on or comes back into range of your
network. This check takes only a second and ensures that all
handsets in your network are kept up-to-date with the same
version of software.
Syslog Server IP The IP address of the syslog server. See the
Diagnostic Tools section for more information.
Time Server IP The IP address of the time server.
SVP Server IP The IP address of the SpectraLink 8000 SVP Server. If
using Static IP configuration, this is simply the IP address of the
SpectraLink 8000 SVP Server. Note that the SpectraLink 8000 SVP
Server must be statically configured to have a permanent IP
address. If DHCP is being used, the handset will try the following,
in order: the DHCP option 151, then a DNS lookup of
“SLNKSVP2” if the DHCP options 6 (DNS server) and 15 (Domain
Name) are configured.
SIP TFTP Server IP The IP address of a TFTP server on your network,
which holds SIP configuration files. In static mode, this parameter
must be configured with an IP address. In DHCP mode, the SIP
TFTP server may be specified by defining the address on the DNS
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
server for the name “siptftp” If this is not defined, the address
specified in option 66 or the boot server/next server (siaddr) field
will be used. See Chapter 4, SIP Integration Factors.
OAI Server IP The IP address of the SpectraLink 8000 OAI
Gateway. If using Static IP configuration, this is simply the IP
address of the SpectraLink 8000 OAI Gateway. If DHCP is being
used, the handset will try the DHCP option 152.
SSID
Enter the SSID.
WLAN Settings
Select between Custom and CCX modes. The Custom mode allows
explicit control of all of the security and QoS settings. Using CCX
mode automatically enables the CCXv4 features and functions, with
only the 802.1X mechanism needing to be selected.
Custom – Security
Handset security setting should match exactly the settings in your
APs. Consult the VIEW Configuration Guide for the APs installed in
your facility for information on which of the security methods are
certified.
Encryption keys, Username and Password display as they are
entered. For security reasons, these items will not display when a
user returns to the Administration menu.
*NONE disables any 802.11 encryption or security authentication
mechanisms.
WEP (Wired Equivalent Privacy) is a wireless encryption protocol that
encrypts data frames on the wireless medium allowing for greater
security in the wireless network. If WEP is required at this site, you
must configure each handset to correspond with the encryption
protocol set up in the APs. Select the entries from the options below to
enable the handset to acquire the system.
Authentication
Select either Open System or Shared Key.
WEP Enable/Disable
Select either Enable WEP or Disable WEP.
Key Information
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Default Key Enter the key number specified for use by the
handsets. This will be 1 through 4.
Key Length Select either 40-bit or 128-bit depending on the key
length specified for use at this location.
Key 1-4 Scroll to the key option that corresponds to the Default
Key that was entered above. Enter the encryption key as a
sequence of hexadecimal characters. (Use the 2 and 3 keys to
access hexadecimal digits A through F.
Rotation Secret
This is used for proprietary WEP key rotation. Refer to your
custom document if this feature is supported in your system.
Cisco FSR (Fast Secure Roaming) FSR is designed to minimize call
interruptions for SpectraLink 8020/8030 Wireless Telephone users as
they roam throughout a facility. Cisco FSR requires specific
configuration of the Cisco APs in your site. See your Cisco
representative for detailed documentation on configuring the APs
and other required security services on the wired network. To
configure Cisco FSR on a handset, you must enter a Radius Server
username and password into each handset.
Username
Enter a username that matches an entry on the RADIUS server.
Usernames are alphanumeric strings, and can be entered using the
alphanumeric string entry technique.
Password
Enter the password that corresponds to this Username.
WPA2-Enterprise
The Authentication setting can select either *EAP-FAST or PEAP as the
authentication method for RADIUS server such as those from
Cisco, Microsoft, or Juniper.
Fast Handoff allows the use of either *CCKM or OKC. These
mechanisms allow a phone to quickly and securely roam between
APs with a minimum disruption of audio.
Username: Enter a username that matches an entry on your
RADIUS server. Alphanumeric strings can be entered using the
alphanumeric string entry technique.
Password: Enter the password that corresponds to this username.
The Delete [PAC/Cert.]: option removes expired credentials from the
phone. When the authentication method is EAP-FAST the PAC
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
on the phone is deleted. If the RADIUS server has enabled
“anonymous in-band PAC provisioning”, then the phone will
automatically re-acquire these credentials from the RADIUS
server over the air. When the authentication method is PEAP or
EAP-FAST manual provisioning, the credential on the phone is
deleted and a new one needs to be downloaded through the HAT.
See additional details in WPA2 Enterprise PEAP Certificate
Enrollment and EAP-FAST Manual PAC Provisioning section later in
this chapter.
Custom – QoS
SVP mode uses the SVP Server to provide enterprise-grade QoS.
DSCP tags are used to change the priority settings for various
classes of packets as they are transmitted to the network from the
Wireless Telephone. Default values are given but may be
overwritten: WT in call = 46, WT standby = 26, Other = 0.
Wi-Fi Standard QoS mode uses standards-based traffic controls for QoS,
instead of the SVP Server.
DSCP tags are used to change the priority settings for various
classes of packets as they are transmitted to the network from the
Wireless Telephone. Default values are given but may be
overwritten: Voice = 46, Control = 26, Other = 0.
Admission Cntrl is used to enable and disable the use of WMM
Admission Control by the handset for the AC_VO and AC_VI
access categories. If the WLAN is using WMM Admission
Control, the handset should be set to *Mandatory. If the WLAN is
not using WMM Admission Control, the handset should be set to
*Optional. See the Best Practices Guide for Deploying SpectraLink
8020/8030 Wireless Telephones for a detailed explanation of the use
of WMM Admission Control.
CCX
CCX settings configure the handset for operation as a CCX V4
certified client.
WPA2-Enterprise
The Authentication setting can select either *EAP-FAST or PEAP as the
authentication method for RADIUS server such as those from
Cisco, Microsoft, or Juniper.
Note that for Fast Handoff, the only selection available is *CCKM.
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Username: Enter a username that matches an entry on your
RADIUS server. Alphanumeric strings can be entered using the
alphanumeric string entry technique.
Password: Enter the password that corresponds to this username.
The Delete [PAC/Cert.]: Option removes expired credentials from
the phone. When the authentication method is EAP-FAST the
PAC on the phone is deleted. If the RADIUS server has enabled
“anonymous in-band PAC provisioning”, then the phone will
automatically re-acquire these credentials from the RADIUS
server over the air. When the authentication method is PEAP or
EAP-FAST manual provisioning, the credential on the phone is
deleted and a new one needs to be downloaded through the HAT.
See additional details in WPA2 Enterprise PEAP Certificate
Enrollment and EAP-FAST Manual PAC Provisioning section later in
this chapter.
QoS – DSCP tags are used to change the priority settings for various
classes of packets as they are transmitted to the network from the
Wireless Telephone. Default values are given but may be overwritten:
Voice = 46, Control = 26, Other = 0.
Regulatory Domain/802.11 Config/Transmit Power
Regulatory domain, 802.11 configuration and transmit power are
interdependent. See Appendix A: Regulatory Domains for regulatory
domain setting specifications. Polycom recommends that you check
with local authorities for the latest status of national regulations for
both 2.4 and 5 GHz wireless LANs. A regulatory domain must be
selected in order for the handset to operate. There is no default
setting.
FCC requirements dictate that the menu for changing the regulatory
domain be available by password, which in our case is the LINE key.
Press LINE and then navigate to the desired domain. Press NavOK to set
the domain.
01 - North America
02 – Europe
03 – Japan
04 – Singapore
05 – Korea
06 – Taiwan
07 – Hong Kong
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
08 – Mexico, India
802.11 config
Once the regulatory domain is set, the 802.11 Config modes are
displayed. Only one may be chosen. 802.11(b & b/g mixed) is the
default. Press NavOK to set the mode. If the mode has subbands, the
Subband list will open. If the mode does not have subbands, the
Transmit Power list will open.
Use g only mode if all of your infrastructure and client devices will
use only 802.11g. The handsets will operate up to 54 Mb/s in this
mode. If any 802.11b capable clients or infrastructure are used in
your wireless LAN then do not use g only mode, instead use 802.11b
and b/g mixed mode for optimum performance.
Use b & b/g mixed if some of your infrastructure components only
understand 802.11b. The handsets will operate up to 11 Mb/s.
Subbands have not been established for the b and b/g mixed or the
g only mode at this writing. Provisions are made in the software to
accommodate these ranges once established. Newly added
subbands may not appear in the Admin menu table above.
Subband
Once a mode is set the subband list will display, if applicable.
Only those ranges which are allowed in the set regulatory domain
and that pertain to the set mode are displayed. Note that for
802.11a the bands labeled DFS will vary depending on the set
regulatory domain. Multiple subbands may be set. Navigate to the
desired subband and set with NavOK. The Transmit Power menu
will open. Once the Transmit Power setting is done, you will be
returned to the subband list.
To deselect a subband, navigate to it and press NavOK.
Once the subband settings are as desired, press the Done softkey to
exit to the Network Setup menu.
Transmit power
For subbands: The Transmit Power list opens when NavOK is
pressed from the Subband menu. A transmit power setting is
required for each subband. Only one level may be set per
subband. Only those power levels which apply to the
regulatory domain and 802.11 mode are listed. Navigate to the
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SpectraLink Wireless Telephone Configuration
desired level and press NavOK to set and return to the subband
list. Another subband may be selected which repeats the
process.
If the highlighted power transmit level is legal on all of the
subbands for the set mode, an All softkey will appear. Press the
All softkey to apply that level to all subbands and return to the
subband menu where all subbands will now be selected. All
overrides any previously set power transmit levels.
Without subbands: When the 802.11 mode has no subbands, the
Transmit Power list opens when NavOK is pressed to set the
mode. Only those power levels which apply to the domain
and 802.11 mode are listed. Navigate to the desired level and
press NavOK. This sets the transmit power level and exits the
Regulatory Domain menus. The Network Setup menu will again
display.
Note that the power setting selected specifies the maximum
for that band/subband. When Transmit Power Control (TPC)
is enabled in the infrastructure, the AP may instruct the
handset to use a lower value to match its own transmit power.
Diagnostics
Run Site Survey
The Site Survey mode is activated by selecting this option. The site
survey starts running immediately upon selecting this option. See the
Diagnostic Tools section for more information about site survey.
Diagnostics Mode
Diagnostics can be enabled or disabled. See Chapter 8 Diagnostic Tools,
section Diagnostics Enabled for a detailed explanation of the Diagnostics
mode options.
Syslog Mode
See Chapter 8 Diagnostic Tools, section Syslog Mode for a detailed
explanation of the Syslog mode options.
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Error Handling Mode
The Error Handling mode determines how the handset will behave
when an error occurs. The Halt on Error option will cause the handset
to stop operating if an error message is received. Unless the error is a
fatal one, normal operation may be resumed by power-cycling the
handset. The Restart on Error option will cause the handset to make
every effort to reboot quietly and quickly to standby mode. In either
scenario, a call in progress will be lost. Restart On Error should be used
unless specific error conditions are being investigated.
Error detail may be shown on the display, captured by the syslog
server and may also be available for downloading with the Handset
Administration Tool. An error memory dump can be taken and sent
to Customer Service for escalation and analysis.
Restore Defaults
The Restore Defaults option will set all user and administrative
parameters except Telephony Protocol to their factory defaults.
Demos
The Graphics Demo option starts a demonstration of the handset’s OAI
graphical capabilities immediately upon selection.
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SpectraLink Wireless Telephone Configuration
Admin Menu Default Table
When the Restore Defaults option is selected, administrative parameters
will be reset to their factory defaults as shown in the table below. The
Telephony Protocol setting will not change. User parameters will be
reset per the user-defined preferences default settings table in the next
section.
Menu option
Setting
Sub-option
Sub-sub-option
Default
PTT/Emerg. Button
Emergency Dial
Disabled
PTT
Disabled
Phone Config
[if enabled]
Allowed
Channels
[all]
Name Channels
[None set]
Priority Channel
Disabled
Time Zone
GMT
Daylight Saving
DST No Adjust
Password
Enabled
Change Password
[n/a]
SIP Registration
[None set]
Clear Regist.
[n/a]
OAI
Enabled
Location Service
Network Config
RTLS
Disabled
Transmit Interval
10 minutes
Location Server IP
[None set]
ELP Port
8552
IP Addresses
Use DHCP
SSID*
[None set]
WLAN Settings
Custom/Security
Custom/QoS
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None
WEP Key Length
40 bit
QoS (Mode)
SVP
QoS (DSCP
tabs)
WT in call = 46
WT standby = 26
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Menu option
Setting
Sub-option
Sub-sub-option
Default
Other = 0
Cisco FSR
Username
[none set]
Password
Reg. Domain*
Diagnostics
[none set]
802.11 mode
b & b/g mixed
Transmit Power
30 mW (15 dBm)
Run Site Survey
[n/a]
Diagnostics
Disabled
Syslog Mode
Disabled
[Error Handling
Mode]
Restart on Error
*Minimum requirements for functionality after Restore Defaults:
Set SSID to an available AP and set Regulatory Domain to 01.
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SpectraLink Wireless Telephone Configuration
User-Defined Preferences
The SpectraLink 8020/8030 Wireless Telephone features a
configuration menu (“Config menu”) that is available to the user to
configure user preferences and display handset information. The
Config menu is opened by pressing the Cfg softkey from standby
mode. See the SpectraLink 8020/8030 Wireless Telephone and Accessories
User Guide.
Config Menu
Config menu
2nd level
3rd level
4th level
5th level
6th level
Ring
Cadence
Off
PBX
Continuous
Short Pulse
Long Pulse
Ring Tone
Tones 1-10
Ring Volume
Volume
Lock Keys
User Profiles
Silent
Vibrate
Loud
Soft
Custom
Set as Active
Ring Settings
Telephone Ring
Message Alert 1
Message Alert 2
„„„„„„„
Vibrate
Cadence
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Off
PBX
Continuous
Short Pulse
Long Pulse
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Config menu
2nd level
3rd level
4th level
Noise Mode1
Normal
5th level
6th level
High
Severe
Ring in
Headset
Ring in
Speaker
Warning Tones
Disable/Enable
Key Tones
Disable/Enable
PTT
Disable/Enable
Phone
Settings
Keypad Autolock
Disable
5 Seconds
10 Seconds
20 Seconds
Display Contrast
Set Contrast
Use Hearing Aid
Use No Hearing
Aid
Play Startup
Song
Inhibit Song
Predial
Disable/Enable
1
High and Severe noise modes increase microphone, speaker, and ring volume settings above Normal mode baseline. All measures are
approximate.
High
Severe
48
Microphone
+12dB
In-ear speaker
+6dB
Ring volume
+3dB
+18dB
+12dB
+6dB
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SpectraLink Wireless Telephone Configuration
Config menu
2nd level
3rd level
Push-to-talk *
Default Channel
Channel 1
4th level
5th level
6th level
….
Channel 24
Subscribed
Channels
Channel 1
Channel 2
Channel 3
….
Channel 24
PTT Audio
Volume
Audio Volume
PTT Tone
Volume
Tone Volume
„„„„„„„
„„„„„„„
PTT Vibrate
Disable/Enable
System Info
Phone IP
Address
Alias IP Address
SVP IP Address
OAI IP Address
Firmware
Version
Emergency Dial *
Emergency
Number
Emergency
Name
* Push-to-talk and Emergency Dial only appear if enabled.
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Default settings
The profile options on the standby menu may be reset to their default
values by the Restore Defaults option in the Admin menu. These are the
default settings:
Setting/profile
Silent
Vibrate
Soft
Loud
Custom
Ring Cadence
Off
Off
PBX
PBX
PBX
Tone 1
Tone 1
Tone 1
Tone 1
Tone 1
1
1
3
7
5
Off
PBX
Off
Off
PBX
Ring Delay
0
0
0
0
5
Noise Mode
Normal
Normal
Normal
Normal
Normal
Headset/Speaker
Speaker
Speaker
Speaker
Speaker
Speaker
Key Tones
Off
Off
On
On
On
Warning Tones
Off
Off
Off
Off
Off
Push-to-talk
Off
Off
On
On
On
PTT Vibrate
Disabled
Disabled
Disabled
Disabled
Disabled
Ring Tone
Ring Volume
Vibrate Cadence
Emergency Dial
On
On
On
On
On
Push-to-talk must be enabled by the system administrator before it can be activated by the user. If it is not
enabled, then it will not appear on the Config menu and will not be “On” for any profile.
PTT Vibrate is available only when Push-to-talk has been enabled by the system administrator.
Emergency Dial must be enabled by the system administrator. If enabled, it will be “On” (or available for
use) in every profile.
WPA2 Enterprise PEAP Certificate Enrollment and EAP-FAST Manual
PAC Provisioning
The Handset Administration Tool (HAT) is used for enrolling a
handset with a PEAP certificate or manually provisioning PAC files.
PEAP
The Handset Administration Tool (HAT) is used for enrolling a
handset with a PEAP certificate in DER format. Only the DER
certification format is supported. All other certificate formats need to
be converted into the DER format prior to enrolling the handset.
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SpectraLink Wireless Telephone Configuration
Choose the Certificate tab and use the file browser to identify the
certificate to be loaded. Once chosen, HAT will perform a
rudimentary check on the file to make sure the format is DER and that
the certificate date is valid. If these tests pass, HAT will indicate that
it is valid and enable the Enroll button. Click Enroll to install the
certificate onto the handset.
The screen below shows a valid certificate that has been identified
with the file browser.
The screen below shows a certificate chosen with the file browser, but
found to be invalid because it has expired.
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
EAP-FAST
For EAP-FAST, HAT is also used for manually provisioning a handset
with a Protected Access Credential (PAC). Choose the PAC file with
the file browser. The user will be prompted to enter the password
used to generate the PAC as part of its validation process. Once the
PAC is considered to be valid, the Provision button will be available for
installing the PAC onto the handset.
The screen below shows a valid PAC identified with the file browser
after a valid password has been entered.
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SpectraLink Wireless Telephone Configuration
The two screens below show the result of entering the wrong
password.
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
If anonymous in-band PAC provisioning is enabled on the RADIUS
server, then it is not necessary to download PAC files through HAT.
The phone will automatically re-acquire credentials from the RADIUS
server over the air.
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3
Software License and Protocol
Management
SpectraLink 8020/8030 Wireless Telephones support a number of
different IP protocol integrations. All SpectraLink 8020/8030 Wireless
Telephones are shipped from Polycom with a generic software load
that allows them to associate to a wireless LAN and download
functional software from a TFTP server. The handsets will not
function properly without downloading appropriate software.
The following details the process to properly configure SpectraLink
8020/8030 Wireless Telephones and download software via over-theair file transfer.
Minimum System Requirements
•
A wireless LAN must be properly configured and operational
through the use of 802.11a/b/g/n wireless APs. Consult the
VIEW Configuration Guide for the appropriate make/model of
WLAN.
•
A TFTP must be available on the network in order to load the
appropriate software into the handsets. The current handset
software must be installed in the proper TFTP download
directory.
•
If SVP is used for QoS, the SVP Server must be installed and
properly configured.
•
Software versions required, if SVP and/or OAI are used:
•
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Component
SpectraLink 8000 SVP Server
Version
17x.034 or higher
OAI Server MOG 600
54.032 or higher
OAI Server MOG 700
82.019 or higher
If Wi-Fi Standard QoS is used, then each AP must be configured
for such features as WMM-Power Save; WMM-Admission
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Control; proper EDCA parameters; DSCP mapping for voice and
control traffic; call admission control and Proxy ARP. Consult the
appropriate VIEW Configuration Guide for these settings.
56
•
If WPA2-Enterprise is used, then all portions of the Public Key
Infrastructure (PKI) need to be installed and configured properly
in order acquire the network.
•
An approved SIP call server must be installed and operational on
the LAN. A complete list of approved product can be found on
the Polycom website at
http://www.polycom.com/support/voice/wifi/feature_lists/spectralink_8020_8030_sip_featurematrix.html
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Software License and Protocol Management
Minimum Configuration Process
1. Download the latest SpectraLink 8020/8030 Wireless Telephone
IP software from
http://www.polycom.com/usa/en/support/voice/wifi/spectralink_8030_wireless.html.
2. Load the latest version of the SIP code and place it on the TFTP
server and ensure the TFTP server is started. The slnk_cfg.cfg is
downloaded first by the phone, and defines the SIP code files that
will be subsequently downloaded. Be sure to use the slnk_cfg.cfg
file that comes with the latest version of the SIP code and do not
change the order of the files within the slnk_cfg.cfg.
The following six files are included in the typical SIP software
package.
Description
Filename
Configuration file
PHINTL (language translation)
USB downloader
Over-The-Air Downloader (OTADL)
OTADL Shim
Functional (telephony protocol)
slnk_cfg.cfg
pi1400sp.bin
pd14udsp.bin
pd14odsp.bin
pd14shim.bin
pd14csp.bin
3. Use the Handset Administration Tool to set up the configuration
of each handset to meet all essential requirements. If not using the
Handset Administration Tool, ensure the following parameters
are correctly set in the Admin menu for each handset: See
Chapter 2 SpectraLink 8020/8030 Wireless Telephone Configuration
for detailed configuration instructions.
— If statically assigning IP addresses, ensure that the Phone IP,
Subnet Mask, and Default Gateway information are accurate. If
using a DHCP Server, ensure that the DHCP option is set.
— Ensure the handset has properly configured SSID and Reg
Domain information.
— Ensure the Telephony Protocol menu option is set to 36. This
ensures the handset will check for the proper SRP files each
time it powers on.
— Ensure security settings are properly programmed.
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
4. Configure handset security settings to match AP configuration
and RADIUS server settings. If WPA2-Enterprise security is used,
credentials will need to be installed onto the handset. For EAPFAST, the PAC file needs to be provisioned and for PEAP the
handset will need to be enrolled with a certificate (requires use of
the HAT). See the WPA2 Enterprise PEAP Certificate Enrollment and
EAP-FAST Manual PAC Provisioning section in this guide for
details.
5. Configure the QoS handset mode to match the AP and site QoS
plan. Follow the VIEW Configuration Guide for the appropriate
make/model of WLAN.
6. Power cycle the handset.
7. The SIP code will now download to the handset. The status bar
will increment fully across the display for each function that is
being performed in the download process. Upon completion of
the update process, the handset will re-boot with the new
firmware.
During the second download evolution, the handset receives code
from the TFTP server for system configuration and for its own
settings. Once this second evolution is complete, the handset is ready
to use.
For future software upgrades, simply update the files that are stored
on the TFTP server. Each time the handset is powered on, it will
check with the TFTP server to ensure it has the proper software
version.
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4
SIP Integration Factors
CODECs
The SpectraLink 8020/8030 Wireless Telephones are compatible with
the G.711µ-law, G.711a-law and G.729 codecs. There is no setting
required on the handset.
DHCP
Dynamic Host Configuration Protocol (DHCP) is a standardized
protocol that enables clients to be dynamically assigned with various
configuration parameters, such as an IP address, subnet mask, default
gateway, and other critical network configuration information. DHCP
servers centrally manage such configuration data, and are configured
by network administrators with settings that are appropriate for a
given network environment. The handset will use the following
DHCP options if DHCP use is enabled:
Option
1
Meaning
Subnet mask
3
Default gateway
6
DNS server
7
Syslog server logging
15
Domain name
66
TFTP server
151
SpectraLink 8000 SVP Server
152
SpectraLink 8000 OAI Gateway
siaddr
Boot server or next server
DNS
Domain Name System (DNS), an industry-standard protocol, locates
computers on an IP-based network. IP networks rely on numberbased addresses to move information on the network. However, it is
easier to remember user-friendly names than number-based
addresses, so it is necessary to translate user-friendly names into
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
addresses that the network can recognize. The handset will use DNS
to automatically translate names into IP addresses for the TFTP server
and SpectraLink 8000 SVP Server.
In DHCP mode, the SIP handset will use DNS to look up an address
for the logical name “siptftp” to locate the SIP TFTP file server. If this
logical name is undefined, then the address specified by option 66 or
the boot server/next server (siaddr) field is used for the SIP TFTP
server.
Sample DHCP Server Configuration File
A sample DHCP server configuration file is illustrated below. Please
note that this is only a sample and will not work on your system as
written here. In addition, this file is specific to the ISC DHCP server.
Your configuration files must be locally programmed according to
your site requirements.
dhcpd.cfg
# /etc/dhcpd.conf
#
# Sample configuration file for ISC dhcpd
#
# Type "man dhcp-options" at prompt to get help for these options.
#
# Global parameters start at beginning of file.
# If this DHCP server is the official DHCP server for the local
# network, the authoritative directive should be uncommented.
# This means the DHCP server will send DHCPNAK messages to misconfigured
# clients.
authoritative;
# Use local address if you want the DHCP server to listen for DHCP requests
# on a specified address, rather than requests send to all addresses.
local-address 192.168.0.1;
# define the default gateway / router option 3
option routers 192.168.0.1;
# define the DNS server(s) option 6
option domain-name-servers 192.168.0.1;
# define the SYSLOG server option 7
option log-servers 192.168.0.1;
# define the domain name option 15
option domain-name "polycom.com";
# define the SVP server option code 151 as an IP address.
option svp-server code 151 = ip-address;
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SIP Integration Factors
# define the OAI server option code 152 as an IP address.
option oai-server code 152 = ip-address;
# This should be the length in seconds that will be
# assigned to a lease if the client requesting the lease does not ask
# for a specific expiration time. 86400 seconds is 1 day.
default-lease-time 86400;
# This should be the maximum length in seconds that will
# to a lease.
max-lease-time 86400;
be
assigned
# minimum lease time of 10 minutes
min-lease-time 600;
#
#
#
#
#
You can declare a class of clients and then do address allocation
based on that.
The example below shows a case where all clients
in a certain class get addresses in the range 192.168.0.100 to 192.168.0.149,
and all other clients get addresses in the range 192.168.0.150 to
192.168.0.199.
class "SpectraLinkPhones" {
match if substring (option vendor-class-identifier, 0, 11) = "SpectraLink";
}
# subnet definition also sets netmask option 1
subnet 192.168.0.0 netmask 255.255.255.0 {
# DHCP lease pool for Polycom SpectraLink phones
pool {
allow members of "SpectraLinkPhones";
range 192.168.0.100 192.168.0.149;
# define the siaddr / next server field as the alternative TFTP server address
next-server 192.168.0.1;
# define the NTP server option 42
option ntp-servers 192.168.0.1;
# define the primary TFTP server address option 66
option tftp-server-name "192.168.0.1";
# define the SVP server address option 151 if using SVP QoS
option svp-server 192.168.0.5; # option 151
# define the OAI server address option 152
option oai-server 192.168.0.6; # option 152
}
# DHCP lease pool for other devices
pool {
deny members of "SpectraLinkPhones";
range 192.168.0.150 192.168.0.199;
next-server 192.168.0.1;
option tftp-server-name "192.168.0.1";
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5
Programming the SIP Handset
Features
In order for the handset to function in the SIP environment, it
downloads two files from the root directory of the SIP TFTP server
during startup. The first file contains generic system information and
is downloaded by every handset during the power-up sequence. A
second file, unique for each handset, is then downloaded. It contains
specific information for each handset such as username, password,
and line appearances. Both of these files must be written locally.
Sample files populated with default parameters are provided with the
software downloaded from the website. We recommend using these
sample files as a starting point and adjusting them according to your
system requirements. See the Sample Configuration Files section below
for the default values.
SIP TFTP Server Configuration Files
The two file types, generic and specific, are identical in format. Any or
all of the configuration information can be contained in either file.
Any information in the specific file that conflicts with the information
in the generic file will take precedence over that in the generic file.
Authentication information will be accepted from both files. For ease
of administration, it is recommended both file types be utilized.
Guidelines
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•
The files are in plain text, US-ASCII. The general form of the
configuration file data is “parameter = value.”
•
The generic filename must be SIP_allusers.cfg.
•
Each specific filename must have the form of SIP_username.cfg
where [username] is as assigned to each individual user by the
system administrator. See Chapter 2 SpectraLink 8020/8030 Wireless
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Telephone Configuration, section The Admin Menu, subsection SIP
Registration.
•
Username parameters are: alphanumeric, no spaces, no
punctuation, case is ignored, 1-16 characters.
•
Generic file information should contain proxy server information
and other SIP system data.
•
Specific file information should contain data specific to each user
such as authentication credentials and line appearance data.
•
Some parameter lines accept more than one value, separated by a
colon or semicolon character as defined in the following table.
•
Any line that begins with a pound sign (#) is ignored.
•
In general space characters are ignored. Space characters may be
included in string values by replacing the space with “%20” or by
enclosing the string in quotes (″).
•
If necessary, other special characters may be included by using a
hexadecimal representation: (%hh) where hh is the representation
of the character.
•
Lines may appear in any order although maintenance may be
simplified by preserving the order in the supplied example file.
Program each of the files according to the following instructions.
The generic file (sip_allusers.cfg)
The generic configuration file provides system information common
to all handsets.
The handset-specific files (e.g. sip_3001.cfg)
The handset-specific configuration file provides specific information
for the handset to identify itself and communicate to other phones.
Each handset must have its own file with a unique filename. You may
use the same parameters as the generic file when programming the
handset files if you wish to override a common setting.
You must configure a unique handset file for each handset being
deployed. Typically each of these files is named with the extension
number or name of the person assigned the handset. For example
John Doe’s handset could have a handset filename of sip_3001.cfg
or sip_JohnDoe.cfg.
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Programming the SIP Handset Features
Proxy server commands
Use the following parameters when programming the configuration
files. See the sample configuration files following this table for more
detailed information.
Parameter
PROXYn_ADDR
Value
xxx.xxx.xxx.xxx:pppp
or
Description
Proxy address. One proxy
address is required but
additional proxy
addresses may be used to
register secondary line
appearances.
Domain name
xxx.xxx.xxx.xxx = IP4
address
pppp = port (optional, 5060
is the default)
proxyname = computer
name (DHCP only)
proxyname:pppp
PROXYn _DOMAIN
Notes
n = 1, 2, 3
Domain served by this
proxy server.
n = 1, 2, 3
DOMAIN = [example:
spectralink.com]
Can be omitted if a specific
proxy domain name is not
defined at the proxy server.
If omitted, defaults to the IP
address of the proxy server
(see above).
PROXYn_TYPE
Asterisk2
NEC (NECSIP)
Mitel
Specify the manufacturer
of each defined proxy
server. See sample files
for vendor notation.
Cisco Call Manager
Express or Unified
Communications
Manager Express.
(CCME)
Used by the handset to
perform proxy-specific
actions based on known
behavior for specific proxy
types. Only the three
shown values may be
used. Omit this parameter
if the value is not one of
these recognized settings.
PingTel
Interactive Intelligence
(ININ)
Toshiba
Cisco Call Manager or
Unified Communcations
Manager (CCM)
3Com
2 For up-to-date Asterisk voicemail commands, go to:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMailMain
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Parameter
PROXYn_
KEEPALIVE_SECS
Value
0 or 10 to 3600
seconds
*default = 0
Description
Notes
Specifies that the handset n = 1, 2, 3
should send keepalives to
the PROXYn server.
The keepalive interval
defaults to ZERO which
disables the feature.
When specified, the
interval can be set to zero
or to 10 to 3600 seconds.
If a keepalive fails to get a
response within the SIP 32
second timeout, then
keepalives are terminated
until the next successful
registration to the proxy.
Not used in
8020/8030
PROXYn_
KEYPRESS_2833
See example file.
*enable
disable
*default (in example
file)
PROXYn_
KEYPRESS_INFO
enable
disable
Controls generation of in- n = 1, 2, 3
stream RFC2833
formatted key press
events. Match what is
required by your PBX. See
RFC2833. See config file Must be disabled for OAI.
example for more
information.
Controls generation of SIP n = 1, 2, 3
INFO requests to the SIP
server for keypress
events. Normally enabled. Must be enabled for OAI.
PROXYn_HOLD_IP0 enable
disable
Controls setting of the
media stream IP
destination address to 0
(zero) when a call is put
on hold. See config file
example for more
information.
n = 1, 2, 3
PROXYn_PRACK
enable
disable
Enables reliable
provisional responses to
INVITE requests
n = 1, 2, 3
PROXYn _
MAIL_SUBSCR
[email protected] Contact to whom the
handset should subscribe
or
for mail notification. See
config file example for
more information.
n = 1, 2, 3
sip:name@domain
Domain where the mail
resides3.
3
Use for compatibility with
older SIP servers that may
not recognize newer
stream attribute
parameters for HOLD
status.
name = mail server contact
name.
xxx.xxx.xxx.xxx = IP4
address.
For mail notifications, in general you will need to define only one contact parameter for each proxy. If the proxy
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Parameter
PROXYn_
MAIL_ACCESS
PROXYn _
MAIL_NOTIFY
Value
Description
[email protected] Contact to whom the
handset should invite to
access the mail center.
or
This is the main voicemail
dial number.
Notes
n = 1, 2, 3
sip:name@domain
Domain where the mail
resides.
n = 1, 2, 3 4
name = mail server contact
name.
xxx.xxx.xxx.xxx = IP
address
[email protected] Contact from whom mail
notification originates. See name = mail server contact
config file example for
name. 5
or
more information.
xxx.xxx.xxx.xxx = IP
address.
sip:name@domain
PROXYn_
REREG_SECS
35 seconds to
3600 seconds
Not used in
8020/8030
*default 3600 seconds
The requested expiration
interval in REGISTER
request messages.
Domain where the mail
resides.
n = 1, 2, 3
If the server has a lower
maximum setting or a
higher minimum setting
than that requested, the
server response takes
precedence.
See example file
PROXYn_FAILOVER xxx.xxx.xxx.xxx:pppp
IP_ADDRESS
or
proxyname:pppp
The IP address of the
Interactive Intelligence
failover call server.
n = 1, 2, 3
xxx.xxx.xxx.xxx = IP4
address
pppp = port (optional, 5060
is the default)
proxyname = computer
name (DHCP only)
server automatically creates and renews subscriptions when the handset registers, then only the
PROXYn _MAIL_NOTIFY contact need be specified. If the handset must subscribe to a particular contact to get mail
notification, then only the PROXYn_MAIL_SUBSCR contact needs to be specified.
4
Values as above.
For Inter-Tel SIP servers, both PROXYn_MAIL_SUBSCR and PROXYn_MAIL_NOTIFY parameters must be set in
the configuration file.
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Parameter
AUTH
[see warning note
below]
Value
username;password
Description
Credentials.
In general credentials are
needed for each
registered line.6.
Notes
username = The dial
number or string that
identifies the line
appearance. Generally an
extension or phone
number.
password = a secure
password created by the
system administer which
enables a handset to
register and/or function.
CODECS
LINEn
LINEn _PROXY
e.g. g711u, g711a,
g729
Comma-separated list of
Defaults to “g711u, g711a”.
supported codecs in order if either is omitted it will be
of preference.
added to the end of the list.
username
The dial # or name.
n = 1, 2, 3, 4, 5
All LINEn user names
should be unique for a
given LINEn_ PROXY.
This may be enforced in
future software revisions.
The registered contact
becomes:
sip:username@domain
SIP proxy server for this
line.
n = 1, 2, 3, 4, 5
codec1, codec2
i
i = the number of the
proxy server 1, 2, 3.
LINEn_PROXY can be
omitted if the line is not to
be registered and you wish
to do direct phone to phone
calls.
LINEn _CALLID
callerid
String that displays at the
far end.
n = 1, 2, 3, 4, 5
callerid = the text that will
display as the caller ID on
the called handset.
WARNING: providing credentials by using the AUTH parameters in the configuration files is a security risk and
should be avoided by entering usernames and passwords in Admin menu or by allowing the user to login at startup
time. Credentials entered here are in plain text and accessible by anyone who can access the TFTP server files.
Credentials stored in the SIP server or in the handsets are protected.
6
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Parameter
FAVORITE
Value
Dialstring;identifier;LIN
En
Description
Phonebook list of numbers
accessible from the
Favorites menu. Favorites
in the allusers file will be
present in the Favorites on
all handsets. See config
file example for more
information.
Notes
Up to 15 entries permitted
which may be divided
between generic and
phone specific files.
Dialstring = complete SIP
URI or local extension #
Identifier = name. if
omitted, the dial string
appears on Favorites
menu.
LINEn = can be omitted if
dialing can be done on any
registered line. Usually
omitted but if present, the
number will be dialed on
the programmed line.
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Sample SIP Configuration Files
Sample SIP configuration files are illustrated below. These files are
available as a download from the software updates site and may be
customized for your application. Please note that these are merely
samples and will not work on your system as written here or as
downloaded. Your configuration files must be locally programmed
according to your site requirements.
SIP_allusers.cfg
# SIP ALL USERS Configuration file example
# Configuration file format example
# Maximum number of characters allowed per line (excluding comments) in this configuration file
is 121.
#-----------------------------------------------------------------------------------------#
#-----------------------------------------------------------------------------------------#
# Codec preference order only. This does not enable/disable codecs. (Optional)
# can be G.711-ulaw, g.711u, G.711U, g711u, G.711U, g729, G729, G.729, g729ab, g729AB
# if g711u is omitted it will be added to end of list.
# if g711a is omitted it will be added to end of list after u.
CODECS = g711u, g711a
#-----------------------------------------------------------------------------------------#
#
# PROXY settings... At least one is required and up to three may be specified.
#
Where one is specicifed, all proxy settings must share the
#
same numbered prefix (i.e. PROXY1_...)
#
# One PROXYn_TYPE (PBX/Call Server) entry is required, additional ones are optional as you
# can register secondary line appearances with other PROXY servers
# Uncomment Your Proxy Type from the list below
#
#PROXY1_TYPE
= Asterisk
#PROXY1_TYPE
= NECSIP
#PROXY1_TYPE
= MITEL
#PROXY1_TYPE
= CCME
#PROXY1_TYPE
= PINGTEL
#PROXY1_TYPE
= ININ
#PROXY1_TYPE
= TOSHIBA
#PROXY1_TYPE
= CCM
#PROXY1_TYPE
= 3COM
#PROXY1_TYPE
= SHORETEL
#PROXY1_TYPE
= DELTAPATH
#-----------------------------------------------------------------------------------------#
#-----------------------------------------------------------------------------------------#
##
##
##
##
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VALUES TO WORK WELL DURING TESTING, THOUGH YOUR SPECIFIC
CONFIGURATION MAY VARY. UNCOMMENT OUT THE FIELDS PRECEEDED BY A
SINGLE # TO ACCEPT THESE VALUES
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## ADDITIONAL INFORMATION ABOUT EACH OF THESE FIELDS AS WELL AS SPECIFIC
## CONFIGURATION EXAMPLES CAN BE FOUND FURTHER DOWN IN THIS FILE
#-----------------------------------------------------------------------------------------#
#-----------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = NECSIP
#PROXY1_ADDR
= 172.29.102.122:5060
# replace the ip address with your Sip Server's
Address
#PROXY1_KEYPRESS_2833 = enable
#PROXY1_KEYPRESS_INFO = disable
#PROXY1_HOLD_IP0 = disable
#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
#PROXY1_DOMAIN = necpolysip:5060
# Replace this with your SIP Domain's name
#PROXY1_CALLID_PER_LINE = disable
#PROXY1_MAIL_ACCESS = 7999
# Put Your Voice Mail Sytem's Pilot Number here
#PROXY1_CONF_IP_ADDRESS
= 172.29.102.131
## For each proxy of type NECSIP, one PROXYn_CONF_IP_ADDRESS is required
## This value is the address of NEC PBX's IP PAD card.
#-----------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = ASTERISK
#PROXY1_ADDR
= 172.29.65.6:5060
Address
#PROXY1_KEYPRESS_2833 = enable
#PROXY1_KEYPRESS_INFO = disable
#PROXY1_HOLD_IP0 = disable
#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
#PROXY1_DOMAIN = axlx.engr.local
#PROXY1_CALLID_PER_LINE = disable
#PROXY1_MAIL_ACCESS = 7999
# replace the ip address with the Asterisk Server's
# Replace this with your SIP Domain's name
# Put Your Voice Mail Sytem's Pilot Number here
#-----------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = MITEL
#PROXY1_ADDR
= 172.29.102.160:5060
#PROXY1_KEYPRESS_2833 = enable
#PROXY1_KEYPRESS_INFO = disable
#PROXY1_HOLD_IP0 = enable
#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
#PROXY1_DOMAIN =
#PROXY1_CALLID_PER_LINE = disable
#PROXY1_MAIL_ACCESS = 7000
# replace the ip address with your PBX's Address
# Put Your SIP Domain's Name Here
# Put Your Voice Mail Sytem's Pilot Number here
#-----------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = CCME
#PROXY1_ADDR
= 172.24.144.107:5060
#PROXY1_KEYPRESS_2833 = disable
#PROXY1_KEYPRESS_INFO = enable
#PROXY1_HOLD_IP0 = enable
#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
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#PROXY1_DOMAIN =
#PROXY1_CALLID_PER_LINE = enable
#PROXY1_MAIL_ACCESS = 7999
# Put Your SIP Domain's Name Here
# Put Your Voice Mail Sytem's Pilot Number here
#-----------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = PINGTEL
#PROXY1_ADDR
= 172.24.144.127:5060
#PROXY1_KEYPRESS_2833 = enable
#PROXY1_KEYPRESS_INFO = disable
#PROXY1_HOLD_IP0 = enable
#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
#PROXY1_DOMAIN = asia.polycom.com
#PROXY1_CALLID_PER_LINE = disable
#PROXY1_MAIL_ACCESS = 101
# replace the ip address with the Pingtel's Address
# Put Your SIP Domain's Name Here
# Put Your Voice Mail Sytem's Pilot Number here
#-----------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = ININ
#PROXY1_ADDR
= 172.24.144.135:8060
port number
#PROXY1_FAILOVERIP = 172.24.144.136:8060
address.
#PROXY1_KEYPRESS_2833 = enable
#PROXY1_KEYPRESS_INFO = disable
#PROXY1_HOLD_IP0 = enable
#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
#PROXY1_DOMAIN =
#PROXY1_CALLID_PER_LINE = disable
#PROXY1_MAIL_ACCESS = 7999
# replace the ip address with the II's Address and
# replace the ip address with the secondary ININ IP
# Put Your SIP Domain's Name Here
# Put Your Voice Mail Sytem's Pilot Number here
#-----------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = TOSHIBA
#PROXY1_ADDR
= 172.29.102.106:5060
#PROXY1_KEYPRESS_2833 = enable
#PROXY1_KEYPRESS_INFO = disable
#PROXY1_HOLD_IP0 = disable
#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
#PROXY1_DOMAIN =
#PROXY1_CALLID_PER_LINE = disable
#PROXY1_MAIL_ACCESS = 290
# replace the ip address with the CIX100's Address
# Put Your SIP Domain's Name Here
# Put Your Voice Mail Sytem's Pilot Number here
#-----------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = CCM
#PROXY1_ADDR
= 172.24.144.103:5060
#PROXY1_KEYPRESS_2833 = enable
#PROXY1_KEYPRESS_INFO = disable
#PROXY1_HOLD_IP0 = disable
#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
#PROXY1_DOMAIN =
#PROXY1_CALLID_PER_LINE = enable
72
# replace the ip address with the CCM's Address
# Put Your SIP Domain's Name Here
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#PROXY1_MAIL_ACCESS = 9999
# Put Your Voice Mail Sytem's Pilot Number here
#-----------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = 3COM
#PROXY1_ADDR
= 172.29.102.190:5060
#PROXY1_KEYPRESS_2833 = enable
#PROXY1_KEYPRESS_INFO = disable
#PROXY1_HOLD_IP0 = enable
#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
#PROXY1_DOMAIN =
#PROXY1_CALLID_PER_LINE = disable
#PROXY1_MAIL_ACCESS = 5000
# replace the ip address with the 3COM's Address
# Put Your SIP Domain's Name Here
# Put Your Voice Mail Sytem's Pilot Number here
#-----------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = DELTAPATH
#PROXY1_ADDR
= 10.240.240.226:5060
Address
#PROXY1_KEYPRESS_2833 = enable
#PROXY1_KEYPRESS_INFO = disable
#PROXY1_HOLD_IP0 = disable
#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
#PROXY1_DOMAIN = axlx.engr.local
#PROXY1_CALLID_PER_LINE = disable
#PROXY1_MAIL_ACCESS = 7999
# replace the ip address with the Asterisk Server's
# Replace this with your SIP Domain's name
# Put Your Voice Mail Sytem's Pilot Number here
#-----------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = SHORETEL
#PROXY1_ADDR
= 172.29.102.212:5060
#PROXY1_KEYPRESS_2833 = enable
#PROXY1_KEYPRESS_INFO = disable
#PROXY1_HOLD_IP0 = disable
#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
#PROXY1_DOMAIN = asia.polycom.com
#PROXY1_CALLID_PER_LINE = disable
#PROXY1_MAIL_ACCESS = 102
# replace the ip address with the Shoretel's Address
# Put Your SIP Domain's Name Here
# Put Your Voice Mail Sytem's Pilot Number here
#######################################################################################
##### This section describes how to add Favorites #####################################
#
Favorites in the allusers file will be present in the favorites on all phones
#
The username can be blank and can include escaped chars
#
Useful features can be included such as call forwarding or dialing voicemail
#FAVORITE = "1234"; "Site Security"
#FAVORITE = "*98"; "Call Forwarding"
#
#
Favorites can also be configured to prompt users to enter more digits. Users will
#
be prompted with characters inside the <>'s.
#
In the below example if the Feature Access Code for call forwarding is #21
#
When the user selects the Favorites option and then Call Forward they would
#
see "Call Forward Enter Destination:"
#
They would then have a chance to enter the fwd destination before pressing the send key
#FAVORITE = #21<Enter Destination:>; "Call Forward"
#
In the example below the user would be prompted "Call Pickup Pickup Number:
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
#
to enter a call pickup number # for a call pickup Feature Access Code starting with *5
#FAVORITE = *5<Pickup Number:>; "Call Pickup"
# If user wants to add a comment for a FAVORITE, it should be added on the next line (not on the
same line where a FAVORITE is defined).
###############################################################################################
##### THIS SECTION EXPLAINS THE PROXY_N PARAMETERS IN MORE DETAIL AND GIVES FORMAT EXAMPLES ####
# One PROXYn_ADDR (PBX/Call Server) entry is required, additional ones are optional as you
# can register secondary line appearances with other PROXY servers
#
#PROXY1_ADDR
= 172.29.102.131:5060
#PROXY2_ADDR
= 172.29.0.140:5060
#-----------------------------------------------------------------------------------------#
# For each proxy of type NECSIP, one PROXYn_CONF_IP_ADDRESS is required. This value is
# the address of NEC PBX's IP PAD card.
#
#PROXY1_CONF_IP_ADDRESS
= 172.29.102.131
#-----------------------------------------------------------------------------------------#
# To test ININ Redundancy PROXY1_FAILOVERIP is required. This value is the secondary IP address
# of the ININ server. If the primary server fails we need to use this address to make the calls.
#
#PROXY1_FAILOVERIP = 172.24.144.136
#-----------------------------------------------------------------------------------------#
#ProxyDomain can be omitted if a specific proxy domain name is not defined at the
# proxy server. If omitted, the ProxyDomain defaults to the IP address of the
# proxy server.
# (below are examples of different ways to specify a domain)
#PROXY1_DOMAIN = plcmengr.com
#PROXY1_DOMAIN = 10.0.0.138
#PROXY1_DOMAIN = axlx.engr.local
#-----------------------------------------------------------------------------------------#
# PROXYn_MAIL_SUBSCR is who we should subscribe to for mail center notifications
# This is needed only if the user is not subscribed automatically at registration.
# It is almost never required in current versions of Asterisk to specify this.
# If you are using Asterisk (non-business edition) before v1.2, this is necessary.
# This example is actually specific for a line number:3001
#PROXY1_MAIL_SUBSCR = sip:[email protected]
#-----------------------------------------------------------------------------------------#
# PROXYn_MAIL_ACCESS is the main voicemail dial number
# Examples:
#PROXY1_MAIL_ACCESS = 7999
# PROXY1_MAIL_ACCESS = sip:[email protected]
# PROXY1_MAIL_ACCESS = [email protected]
# PROXY1_MAIL_ACCESS = 7999
#-----------------------------------------------------------------------------------------#
#PROXYn_KEYPRESS_2833 controls generation
# press events. Normally you want this to
# on your configuration and what you want
# If you are going to do OAI integration,
# The default is disable
#PROXY1_KEYPRESS_2833 = disable
74
of in-stream RFC2833 formatted key
be disabled for Asterisk but it depends
to be able to do.
this must be disabled.
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#-----------------------------------------------------------------------------------------#
#PROXYn_KEYPRESS_INFO controls generation of SIP INFO requests to the SIP server
# for keypress events. Normally you want this to be enabled.
# If you are going to do OAI integration, this must be enabled.
# The default is enable
#PROXY1_KEYPRESS_INFO = enable
#-----------------------------------------------------------------------------------------#
# PROXYn_HOLD_IP0 controls setting of media stream IP destination to 0.0.0.0 when
# a call is put on hold.
# PROXYn_HOLD_IP0 is not required for current versions of Asterisk.
# For older PBXs that require this, set this to enable
#PROXYn_HOLD_IP0 = enable
#-----------------------------------------------------------------------------------------#
# PROXYn_PRACK enables ACK'd provisional responses to INVITE requests. The PRACK
# mechanism will be used if this switch is enabled and the Proxy server specifies
# support for the PRACK mechanism. PRACK is NOT SUPPORTED in current versions of
# Asterisk, but is to be supported on subsequent versions. PRACK should not be
# required on local area networks.
#PROXY1_PRACK = disable
#-----------------------------------------------------------------------------------------#
# PROXYn_KEEPALIVE_SECS specifies that the phone should send Keep Alives to the
# PROXYn server. It can be set to 0 or anywhere from 10 to 3600 seconds.
# The default value (which is applied if no parameter is supplied in this file) is
# 0 which means the phone will not send keep_alives to PROXYn.
#
# If the phone failes to get a keepalive response within the SIP 32 second
# timeout, the keepalives are terminated for that PROXY until the next successful
# registration to the PROXY.
#
# This parameter is REQUIRED on Asterisk Business Edition with the Polycom 8002
# phone. For Polycom 8020/8030 phones, this parameter is required when they are
# operating in WI-FI standard QoS mode.This is because Linux, the OS running Asterisk,
# has a ARP Cache flushing mechanism. This feature serves to refresh the ARP Cache for the 8002
phones.
# For 8020 and 8030 phones, operating in SVP mode ARP Cache is refreshed by the SVP server.
# If the phones are operating in standard based QoS method ARP cache refresh has to be done by
phone.
#
# The value of this parameter for phones should be just less than the
# minimum ARP Cache flushing time on the Linux box running Asterisk. For Redhat
# this minimum flushing time is 15 seconds, so the parameter needs to be set for
# 14 seconds or less. For other Linux Distributions, you may have to determine
# what this minimum value is, however 14 seconds should work fine.
#
# The lower this setting, the more the impact on the phone's standby battery life.
# (8002 phones running version 130.003 and earlier code ignore this parameter if
# present)
# 8020/8030 phones running version 131.017 and earlier code ignore this parameter if
# present)
#PROXY1_KEEPALIVE_SECS=14
#-----------------------------------------------------------------------------------------------#
# PROXYn_REREG_SECS specifies the the re-registration interval for a PROXY. This
# is the requested expiration interval sent in the REGISTER request message. It can
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
# be set to anything between 35 and 3600 seconds.
# The default value (which is applied if no parameter is supplied in this file) is
# 3600 which means the phone will attempt to register every 3580 seconds, if the
# server's response interval doesn't take precedence (see below).
#
# If the server has a lower maximum setting or a higher minimum setting that
# this value then the server's response will take precedence and this parameter
# will be ignored.
#
# The phone will always attempt to re-register with the PROXY 20 seconds prior to
# the expiration interval. Thus, if this parameter is set to 35 seconds, and the
# server's response interval doesn't take precedence, then the phone will attempt
# to re-register every 15 seconds.
#
# The lower this setting, the more the impact on the 8002's standby battery life.
#(8002 phones running version 130.003 and earlier code ignore this parameter if
# present)
#PROXY1_REREG_SECS=3600
#-----------------------------------------------------------------------------------------#
# PROXYn_CALLID_PER_LINE controls sending of same Call-ID header
# field for registrations sent to a particular registrar."
# If this is enable we will send Call-ID headers differently.
# This may be needed for Mitel 3300 Controller software prior to Release 9
#PROXYn_CALLID_PER_LINE = enable
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Sample handset-specific file
# SIP Extension Specific Configuration file format example
# Maximum number of characters allowed per line (excluding comments) in this configuration file
is 121.
#
#
#
#
#
/////////////////////////////////////////////////////////////////////////////////////////////#
////// Items IN THE PROXY SECTION should be in the sip_allusers.cfg file with
//////
items uncommented in this file only for overriding a setting for a
//////
particular user.
/////////////////////////////////////////////////////////////////////////////////////////////#
# Codec preference order only. This does not enable/disable codecs. (Optional)
# can be G.711-ulaw, g.711u, G.711U, g711u, G.711U, g729, G729, G.729, g729ab, g729AB
# if g711u is omitted it will be added to end of list.
# if g711a is omitted it will be added to end of list after u.
#CODECS = g711u, g711a
#-----------------------------------------------------------------------------------------------#
#
# PROXY settings... At least one is required and up to three may be specified.
#
Where one is specicifed, all proxy settings must share the
#
same numbered prefix (i.e. PROXY1_...)
#
# One PROXYn_TYPE (PBX/Call Server) entry is required, additional ones are optional as you
# can register secondary line appearances with other PROXY servers
# Uncomment Your Proxy Type from the list below
#
#PROXY1_TYPE
= Asterisk
#PROXY1_TYPE
= NECSIP
#PROXY1_TYPE
= MITEL
#PROXY1_TYPE
= CCME
#PROXY1_TYPE
= PINGTEL
#PROXY1_TYPE
= ININ
#PROXY1_TYPE
= TOSHIBA
#PROXY1_TYPE
= CCM
#PROXY1_TYPE
= 3COM
#PROXY1_TYPE
= SHORETEL
#PROXY1_TYPE
= DELTAPATH
#-----------------------------------------------------------------------------------------------#
#-----------------------------------------------------------------------------------------------#
##
##
##
##
##
##
FOR EACH PROXY TYPE LISTED ABOVE POLYCOM HAS FOUND THE FOLLOWING
VALUES TO WORK WELL DURING TESTING, THOUGH YOUR SPECIFIC
CONFIGURATION MAY VARY. UNCOMMENT OUT THE FIELDS PRECEEDED BY A
SINGLE # TO ACCEPT THESE VALUES
ADDITIONAL INFORMATION ABOUT EACH OF THESE FIELDS AS WELL AS SPECIFIC
CONFIGURATION EXAMPLES CAN BE FOUND FURTHER DOWN IN THIS FILE
#-----------------------------------------------------------------------------------------------#
#-----------------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = NECSIP
#PROXY1_ADDR
Address
= 172.29.102.122:5060
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# replace the ip address with your Sip Server's
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
#PROXY1_KEYPRESS_2833 = enable
#PROXY1_KEYPRESS_INFO = disable
#PROXY1_HOLD_IP0 = disable
#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
#PROXY1_DOMAIN = necpolysip:5060
# Replace this with your SIP Domain's name
#PROXY1_CALLID_PER_LINE = disable
#PROXY1_MAIL_ACCESS = 7999
# Put Your Voice Mail Sytem's Pilot Number here
#PROXY1_CONF_IP_ADDRESS
= 172.29.102.121
## For each proxy of type NECSIP, one PROXYn_CONF_IP_ADDRESS is required
## This value is the address of NEC PBX's IP PAD card.
#-----------------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = ASTERISK
#PROXY1_ADDR
= 172.29.65.6:5060
Address
#PROXY1_KEYPRESS_2833 = enable
#PROXY1_KEYPRESS_INFO = disable
#PROXY1_HOLD_IP0 = disable
#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
#PROXY1_DOMAIN = axlx.engr.local
#PROXY1_CALLID_PER_LINE = disable
#PROXY1_MAIL_ACCESS = 7999
# replace the ip address with the Asterisk Server's
# Replace this with your SIP Domain's name
# Put Your Voice Mail Sytem's Pilot Number here
#-----------------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = MITEL
#PROXY1_ADDR
= 172.29.102.160:5060
#PROXY1_KEYPRESS_2833 = enable
#PROXY1_KEYPRESS_INFO = disable
#PROXY1_HOLD_IP0 = enable
#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
#PROXY1_DOMAIN =
#PROXY1_CALLID_PER_LINE = disable
#PROXY1_MAIL_ACCESS = 7000
# replace the ip address with your PBX's Address
# Put Your SIP Domain's Name Here
# Put Your Voice Mail Sytem's Pilot Number here
#-----------------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = CCME
#PROXY1_ADDR
= 172.24.144.107:5060
#PROXY1_KEYPRESS_2833 = disable
#PROXY1_KEYPRESS_INFO = enable
#PROXY1_HOLD_IP0 = enable
#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
#PROXY1_DOMAIN =
#PROXY1_CALLID_PER_LINE = enable
#PROXY1_MAIL_ACCESS = 7999
# replace the ip address with the UCME's Address
# Put Your SIP Domain's Name Here
# Put Your Voice Mail Sytem's Pilot Number here
#-----------------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = PINGTEL
#PROXY1_ADDR
= 172.24.144.127:5060
#PROXY1_KEYPRESS_2833 = enable
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#PROXY1_KEYPRESS_INFO = disable
#PROXY1_HOLD_IP0 = enable
#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
#PROXY1_DOMAIN = asia.polycom.com
#PROXY1_CALLID_PER_LINE = disable
#PROXY1_MAIL_ACCESS = 101
# Put Your SIP Domain's Name Here
# Put Your Voice Mail Sytem's Pilot Number here
#-----------------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = ININ
#PROXY1_ADDR
= 172.24.144.135:8060
port number
#PROXY1_FAILOVERIP = 172.24.144.136:8060
address.
#PROXY1_KEYPRESS_2833 = enable
#PROXY1_KEYPRESS_INFO = disable
#PROXY1_HOLD_IP0 = enable
#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
#PROXY1_DOMAIN =
#PROXY1_CALLID_PER_LINE = disable
#PROXY1_MAIL_ACCESS = 7999
# replace the ip address with the II's Address and
# replace the ip address with the secondary ININ IP
# Put Your SIP Domain's Name Here
# Put Your Voice Mail Sytem's Pilot Number here
#-----------------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = TOSHIBA
#PROXY1_ADDR
= 172.29.102.106:5060
#PROXY1_KEYPRESS_2833 = enable
#PROXY1_KEYPRESS_INFO = disable
#PROXY1_HOLD_IP0 = enable
#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
#PROXY1_DOMAIN =
#PROXY1_CALLID_PER_LINE = disable
#PROXY1_MAIL_ACCESS = 290
# replace the ip address with the CIX100's Address
# Put Your SIP Domain's Name Here
# Put Your Voice Mail Sytem's Pilot Number here
#-----------------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = CCM
#PROXY1_ADDR
= 172.24.144.103:5060
#PROXY1_KEYPRESS_2833 = enable
#PROXY1_KEYPRESS_INFO = disable
#PROXY1_HOLD_IP0 = enable
#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
#PROXY1_DOMAIN =
#PROXY1_CALLID_PER_LINE = enable
#PROXY1_MAIL_ACCESS = 9999
# replace the ip address with the CCM's Address
# Put Your SIP Domain's Name Here
# Put Your Voice Mail Sytem's Pilot Number here
#-----------------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = 3COM
#PROXY1_ADDR
= 172.29.102.190:5060
#PROXY1_KEYPRESS_2833 = enable
#PROXY1_KEYPRESS_INFO = disable
#PROXY1_HOLD_IP0 = enable
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#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
#PROXY1_DOMAIN =
#PROXY1_CALLID_PER_LINE = disable
#PROXY1_MAIL_ACCESS = 5000
# Put Your SIP Domain's Name Here
# Put Your Voice Mail Sytem's Pilot Number here
#-----------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = DELTAPATH
#PROXY1_ADDR
= 10.240.240.226:5060
Address
#PROXY1_KEYPRESS_2833 = enable
#PROXY1_KEYPRESS_INFO = disable
#PROXY1_HOLD_IP0 = disable
#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
#PROXY1_DOMAIN = axlx.engr.local
#PROXY1_CALLID_PER_LINE = disable
#PROXY1_MAIL_ACCESS = 7999
# replace the ip address with the Asterisk Server's
# Replace this with your SIP Domain's name
# Put Your Voice Mail Sytem's Pilot Number here
#-----------------------------------------------------------------------------------------#
## FOR PROXY1_TYPE = SHORETEL
#PROXY1_ADDR
= 172.29.102.212:5060
#PROXY1_KEYPRESS_2833 = enable
#PROXY1_KEYPRESS_INFO = disable
#PROXY1_HOLD_IP0 = disable
#PROXY1_PRACK = enable
#PROXY1_REREG_SECS=3600
#PROXY1_KEEPALIVE_SECS=14
#PROXY1_DOMAIN = asia.polycom.com
#PROXY1_CALLID_PER_LINE = disable
#PROXY1_MAIL_ACCESS = 102
# replace the ip address with the Shoretel's Address
# Put Your SIP Domain's Name Here
# Put Your Voice Mail Sytem's Pilot Number here
###############################################################################################
##### THIS SECTION EXPLAINS THE PROXY_N PARAMETERS IN MORE DETAIL AND GIVES FORMAT EXAMPLES ####
# One PROXYn_ADDR (PBX/Call Server) entry is required, additional ones are optional as you
# can register secondary line appearances with other PROXY servers
#
#PROXY1_ADDR
= 172.29.102.131:5060
#PROXY2_ADDR
= 172.29.0.140:5060
#-----------------------------------------------------------------------------------------------#
# For each proxy of type NECSIP, one PROXYn_CONF_IP_ADDRESS is required. This value is
# the address of NEC PBX's IP PAD card.
#
#PROXY1_CONF_IP_ADDRESS
= 172.29.102.131
#-----------------------------------------------------------------------------------------------#
# To test ININ Redundancy PROXY1_FAILOVERIP is required. This value is the secondary IP address
# of the ININ server. If the primary server fails we need to use this address to make the calls.
#
#PROXY1_FAILOVERIP = 172.24.144.136
#-----------------------------------------------------------------------------------------------#
#ProxyDomain can be omitted if a specific proxy domain name is not defined at the
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# proxy server. If omitted, the ProxyDomain defaults to the IP address of the
# proxy server.
# (below are examples of different ways to specify a domain)
#PROXY1_DOMAIN = plcmengr.com
#PROXY1_DOMAIN = 10.0.0.138
#PROXY1_DOMAIN = axlx.engr.local
#-----------------------------------------------------------------------------------------------#
# PROXYn_MAIL_SUBSCR is who we should subscribe to for mail center notifications
# This is needed only if the user is not subscribed automatically at registration.
# It is almost never required in current versions of Asterisk to specify this.
# If you are using Asterisk (non-business edition) before v1.2, this is necessary.
# This example is actually specific for a line number:3001
#PROXY1_MAIL_SUBSCR = sip:[email protected]
#-----------------------------------------------------------------------------------------------#
# PROXYn_MAIL_ACCESS is the main voicemail dial number
# Examples:
#PROXY1_MAIL_ACCESS = 7999
# PROXY1_MAIL_ACCESS = sip:[email protected]
# PROXY1_MAIL_ACCESS = [email protected]
# PROXY1_MAIL_ACCESS = 7999
#-----------------------------------------------------------------------------------------------#
#PROXYn_KEYPRESS_2833 controls generation
# press events. Normally you want this to
# on your configuration and what you want
# If you are going to do OAI integration,
# The default is disable
#PROXY1_KEYPRESS_2833 = disable
of in-stream RFC2833 formatted key
be disabled for Asterisk but it depends
to be able to do.
this must be disabled.
#-----------------------------------------------------------------------------------------------#
#PROXYn_KEYPRESS_INFO controls generation of SIP INFO requests to the SIP server
# for keypress events. Normally you want this to be enabled.
# If you are going to do OAI integration, this must be enabled.
# The default is enable
#PROXY1_KEYPRESS_INFO = enable
#-----------------------------------------------------------------------------------------------#
# PROXYn_HOLD_IP0 controls setting of media stream IP destination to 0.0.0.0 when
# a call is put on hold.
# PROXYn_HOLD_IP0 is not required for current versions of Asterisk.
# For older PBXs that require this, set this to enable
# Try toggling this value if you are having problems with hold
#PROXYn_HOLD_IP0 = enable
#-----------------------------------------------------------------------------------------------#
# PROXYn_PRACK enables ACK'd provisional responses to INVITE requests. The PRACK
# mechanism will be used if this switch is enabled and the Proxy server specifies
# support for the PRACK mechanism. PRACK is NOT SUPPORTED in current versions of
# Asterisk, but is to be supported on subsequent versions. PRACK should not be
# required on local area networks.
#PROXY1_PRACK = disable
#-----------------------------------------------------------------------------------------------#
# PROXYn_KEEPALIVE_SECS specifies that the phone should send Keep Alives to the
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# PROXYn server. It can be set to 0 or anywhere from 10 to 3600 seconds.
# The default value (which is applied if no parameter is supplied in this file) is
# 0 which means the phone will not send keep_alives to PROXYn.
#
# If the phone failes to get a keepalive response within the SIP 32 second
# timeout, the keepalives are terminated for that PROXY until the next successful
# registration to the PROXY.
#
# This parameter is REQUIRED on Asterisk Business Edition with the Polycom 8002
# phone. For Polycom 8020/8030 phones, this parameter is required when they are
# operating in WI-FI standard QoS mode.This is because Linux, the OS running Asterisk,
# has a ARP Cache flushing mechanism. This feature serves to refresh the ARP Cache for the 8002
phones.
# For 8020 and 8030 phones, operating in SVP mode ARP Cache is refreshed by the SVP server.
# If the phones are operating in standard based QoS method ARP cache refresh has to be done by
phone.
#
# The value of this parameter for phones should be just less than the
# minimum ARP Cache flushing time on the Linux box running Asterisk. For Redhat
# this minimum flushing time is 15 seconds, so the parameter needs to be set for
# 14 seconds or less. For other Linux Distributions, you may have to determine
# what this minimum value is, however 14 seconds should work fine.
#
# The lower this setting, the more the impact on the phone's standby battery life.
# (8002 phones running version 130.003 and earlier code ignore this parameter if
# present)
# 8020/8030 phones running version 131.017 and earlier code ignore this parameter if
# present)
#PROXY1_KEEPALIVE_SECS=14
#-----------------------------------------------------------------------------------------------#
# PROXYn_REREG_SECS specifies the the re-registration interval for a PROXY. This
# is the requested expiration interval sent in the REGISTER request message. It can
# be set to anything between 35 and 3600 seconds.
# The default value (which is applied if no parameter is supplied in this file) is
# 3600 which means the phone will attempt to register every 3580 seconds, if the
# server's response interval doesn't take precedence (see below).
#
# If the server has a lower maximum setting or a higher minimum setting that
# this value then the server's response will take precedence and this parameter
# will be ignored.
#
# The phone will always attempt to re-register with the PROXY 20 seconds prior to
# the expiration interval. Thus, if this parameter is set to 35 seconds, and the
# server's response interval doesn't take precedence, then the phone will attempt
# to re-register every 15 seconds.
#
# The lower this setting, the more the impact on the 8002's standby battery life.
#(8002 phones running version 130.003 and earlier code ignore this parameter if
# present)
#PROXY1_REREG_SECS=3600
#-----------------------------------------------------------------------------------------------#
# PROXYn_CALLID_PER_LINE controls sending of same Call-ID header
# field for registrations sent to a particular registrar."
# If this is enable we will send Call-ID headers differently.
# This may be needed for Mitel 3300 Controller software prior to Release 9
#PROXYn_CALLID_PER_LINE = enable
#-----------------------------------------------------------------------------------------------#
# PROXY1_MAIL_SUBSCR is who we should subscribe to for mail center notifications
# This is needed only if the user is not subscribed automatically at registration.
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# It is almost never required in current versions of Asterisk to specify this.
# Mitel, Cisco, Interactive Intelligence, and Pingtel also accept this parameter
# If you are using Asterisk (non-business edition) before v1.2, this is necessary.
# This example is specific for a line number:3001
#PROXY1_MAIL_SUBSCR = sip:[email protected]
#PROXY1_MAIL_SUBSCR = sip:[email protected]
# //////////////////////////////////////////////////////////////////////////////////////////////#
# ////// The administrator should enter the extension specific configuration
# ////// parameters for this extension in this section of the file
# /////////////////////////////////////////////////////////////////////////////////////////////#
# Authentication credentials (Normally not stored in this file for security reasons)
# AUTH = username; password
# For lines on proxies of type NECSIP and SHORETEL, the LINEn_AUTH setting should be used instead
# as each NEC line must respond to any authentication challenge using a single UN/PW pair.
#AUTH = 3001; 3001
# ////// For CCME, the value for LINEn_CALLID cannot have commas or special characters.
# ////// If a comma or a special character is in this command, phone will display an
# ////// error message "Bad Request 'Mal".
#-----------------------------------------------------------------------------------------------#
# Line definitions
# Each definition should have LINEn, LINEn_PROXY and LINEn_CALLID
# LINEn is the dial number
# LINEn_PROXY is the PROXYn server this line should register with, typically
#
defined in sip_allusers.cfg.
# LINEn_CALLID is shown on the standby display of 8020/8030 phones but not 8002
#
or e/h340/i640 phones. The Asterisk Server converts the callID information
#
to alternative forms defined in the Asterisk configuration files for
#
display at the far end of a phone call.
# LINEn_AUTH is a means of associating a particular user-name/Password pair with a specific
line.
# Linen_AUTH method of authentication must be used on PROXY types; NEC and SHORETEL
#
Like AUTH, there is a security risk associated with storing authentication credentials in
this file.
#
alternatively, the LINEn_AUTH setting information can be administered via the SIP Reg menus.
# Up to 5 line definitions can be made for each user
# Line definitions do not necessarily have to have different extensions
# ////// For CCME, the value for LINEn_CALLID shouldn't have commas or special characters.
# ////// If found, the phone will display an error message "Bad Request 'Mal".
LINE1
LINE1_PROXY
LINE1_CALLID
LINE1_AUTH
=
=
=
=
3001
1
User Name 1
3001; 3001
# Two lines may
LINE2
=
LINE2_PROXY
=
LINE2_CALLID =
LINE2_AUTH
=
map to the same extension to allow second incoming calls.
3002
1
User Name 2
3002; 3002
LINE3
LINE3_PROXY
LINE3_CALLID
LINE3_AUTH
3003
1
User Name 3
3003; 3003
#LINE4
=
=
=
=
= 3004
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#LINE4_PROXY
#LINE4_CALLID
#LINE4_AUTH
= 3
= User Name 4
= 3004; 3004
#LINE5
#LINE5_PROXY
#LINE5_CALLID
#LINE5_AUTH
=
=
=
=
3005
2
User Name 5
3005; 3005
#######################################################################################
##### This section describes how to add Favorites #####################################
# Favorite Dialed Number list.
#
You can define up to 15 total entries including any defined in sip_allusers.cfg.
#
You can enclose a string in quotes to allow for spaces.
#
Each favorite can be complete SIP URI
#
Format is:
#
FAVORITE = dial_string; username
#
#
The username can be blank and can include escaped chars.
FAVORITE = 3001; User Name 1
#FAVORITE = 3002; User Name 2
#FAVORITE = 3003; User Name 3
#FAVORITE = 3004; User Name 4
#FAVORITE = 93035551212; User Name 5 Cell
#FAVORITE = 3006; User Name 6
#FAVORITE = 3007; User Name 7
#FAVORITE = 3008; User Name 8
#FAVORITE = 3009; User Name 9
#FAVORITE = 3010; User Name 10
#FAVORITE = 3011; User Name 11
#FAVORITE = 3013; User Name 12
#FAVORITE = 3013; User Name 13
#FAVORITE = 3014; User Name 14
#FAVORITE = 3015; User Name 15
#
#
Favorites can also be configured to prompt users to enter more digits. Users will
#
be prompted with characters inside the <>'s.
#
In the below example if the Feature Access Code for call forwarding is #21
#
When the user selects the Favorites option and then Call Forward they would
#
see "Call Forward Enter Destination:"
#
They would then have a chance to enter the fwd destination before pressing the send key
#FAVORITE = #21<Enter Destination:>; "Call Forward"
#
In the example below the user would be prompted "Call Pickup Pickup Number:
#
to enter a call pickup number # for a call pickup Feature Access Code starting with *5
#FAVORITE = *5<Pickup Number:>; "Call Pickup"
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6
Using the SIP Handset
The Handset Display
When active, the handset screen will display either a call status screen
or one of several menu screens. The call status screen has the
following format:
This example shows two call tabs indicating that two calls are in
progress. The un-selected call tab indicates that we have put another
call on hold. The call-status icon for the selected call indicates that this
call is being dialed. The text indicates the selected call is on line 1,
extension 3001. Enter Number indicates that the handset is ready to
be dialed. Once this call is connected, the connected party’s
information will appear on the third line, and the fourth line contains
help or error messages, as appropriate. The softkeys during this
action offer text editing functions.
System icons
Indicator
Function
The signal-strength icon indicates the strength of the signal and can assist
the user in determining if the handset is moving out-of-range.
The voicemail icon is activated when a new voicemail message is received
if the feature is supported by the phone emulation.
The battery icon indicates the amount of charge remaining in the Battery
Pack. When only one level remains, the Battery Pack needs to be charged.
The speakerphone icon displays when the speakerphone is active.
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Indicator
Function
Up and down arrows are displayed when the menu has additional options
above or below.
Left or right arrows are displayed during editing when the cursor may be
moved left or right.
The Push-to-talk (PTT) ring icon. A PTT call is coming in.
The priority PTT ring icon. A call is coming in on the priority PTT channel.
This call will override any other.
Location Service icon: indicates the Ekahau Real-Time Location System
(RTLS) is enabled.
Locked
[No Service
message]
Locked indicates that the keypad is locked to prevent accidental activation.
Use the Unlk softkey plus the # key to unlock it.
If warning tones are not disabled, an alarm will sound and a descriptive
message displays when the handset cannot receive or place calls. You may
be outside of the covered area. Walk back into the covered area. The inservice tone indicates service is reestablished.
The download icon indicates that the handset is downloading code. This
icon only appears while the handset is running the over-the-air downloader.
It appears to the right of the Signal Strength icon in the same location as the
Voicemail icon.
XXXX
During character entry, Indicates current data entry symbol mode.
Call status icons
Indicator
Function
On-hook icon, Solid when in standby mode to indicate that at least one call
is still active or on hold. Flashing when incoming call is ringing.
Off-hook icon. Solid when a call is being dialed.
Hold icon. Call is on hold
Transfer icon. Call is in the process of being transferred
Audio flowing icon. Audio is flowing both ways on a call.
Audio receive-only icon. Locally muted (flash) or far end hold with no music
on hold.
No audio icon. No audio is flowing. Call is terminating or far end hold with
audio disable.
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Using the SIP Handset
NavOK functions
The NavOK key acts as a fifth softkey with implicit functionality as
follows:
State
Dialing
Answering
Holding
Displaying menu
Displaying call status
Entering login name
or login password
NavOK key function
Place phone call.
Answer a second phone call (same or different line)
Resume audio.
Select the highlighted menu option.
Resume audio on the currently selected call and place previous call on
hold. If the selected call is ringing, the call will be answered.
Save name or password and proceed with startup.
Softkeys
Softkey
<<
Name
Cursor backward
Displayed during…
Entering a dial number.
Press to…
Move the cursor back one position.
>>
Cursor forward
Entering a dial number.
Move the cursor forward in
alphanumeric mode, if the cursor is
at the end of the line, adds a space
character.
Answ
Answer
Incoming call on the selected
line.
Answer the call (equivalent to START
key).
Bksp
Backspace
character
Entering a dial number.
Delete the character prior to the
cursor position.
Back
Back one screen
Displaying a menu.
Exit the menu.
Dial
Dial Call
A dial number is being
entered on the selected line.
Initiate a phone call to the entered
dial number.
End
End Call
An active call on the selected
line.
Terminate the call without going back
to standby mode.
Favr
Favorites
Prior to entering the first
character of a dial number
Activate the Favorites menu.
Fwd
Forward Definition
Prior to entering the first
character of a dial number.
Delete a previously defined forward
destination.
Or initiate definition of a new forward
destination.
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Softkey
Hold
Name
Hold
Displayed during…
In an active call.
Press to…
Place the call on hold. The line status
shows
when the call is on hold or
when audio is flowing.
Msg
Message
Initial dial screen when new
line is selected and a dial
tone is active prior to
entering first character of the
number to be dialed7.
Initiate a call to the specified
message center contact address for
retrieval or administration of
voicemail.
Mute
Toggle muting
In an active call.
Toggle audio transmission to the far
end. The line status shows
when
not muted or
when muted.
OK
OK
Power up registration if
username is not configured
in Admin menu.
Send the username and password to
the SIP server for authorization to
register the handset.
Redl
Redial
Prior to entering the first
character of a dial number.
Redial the last number that was
dialed.
Rej
Reject
Incoming call on the selected
line.
Reject the incoming call. The SIP
server will then redirect the call
elsewhere.
Resm
Resume
In an active call and you
have placed the call on hold
or in standby mode if any call
is on hold.
Resume a call that was previously
placed on hold or that went on hold
when another line was activated.
Save
Save
Entering a dial number as a
forward destination.
Save the dial number as the
forwarding destination for the
selected line.
Symb
Symbols
Entering a username or
password.
Select the set of symbols available
on the keypad while entering data.
Entering the digits of a
number.
7 Appears
only if the PROXYn_MAIL_NOTIFY is configured. A message center contact address must be defined for
the proxy used by the selected line.
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Using the SIP Handset
Menus
Line menu
The Line menu allows you to activate a call on a selected line or to
view the status of lines.
Pressing the LINE key from the active mode displays a menu of line
appearances as programmed in the SIP TFTP configuration file. The
LINE key can be pressed while the handset is in the standby mode to
activate the handset and to activate a new call on the selected line.
The currently selected line is indicated by an asterisk (*). Lines for
which the corresponding proxy server has outstanding new mail are
flagged with plus (+) characters. Lines that should be registered to a
proxy but have failed registration for any reason are displayed in
faded text and are not selectable from the menu.
Exit the LINE display by pressing a line number key to start a new call
on the selected line and put any other call on hold, or by pressing the
END key to exit without starting a new call. Press the More softkey to
page through additional items on the Line menu.
Symbol menu
The symbol menu allows you to change the set of characters available
for data entry through multiple key presses of the dial pad keys.
While dialing a number or entering login information, press the Symb
softkey to view a menu of possible sets of characters that can be
entered using multiple key presses of the dial pad keys. Normally, a
simple numeric mode is selected; selecting other symbol modes
allows convenient access to the complete printable US ASCII
character set. The following table shows what characters are available
through repeated key presses in various symbol modes.
PN: 1725-36038-001-H.doc
Key
1
Number
1
English
1 ; : / \ ! ‘
Number + English
1
Punctuation
@:1
2
2
abc2ABC
2ABCabc
;,2
3
3
def3DEF
3DEFdef
&|`~3
4
4
ghI4GHI
4GHIghi
( ) 4
5
5
jkl5JKL
5JKLjkl
< > 5
6
6
mno6MNO
6MNOmno
{ } 6
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Key
7
Number
7
English
pqrs7PQRS
Number + English
7PQRSpqrs
Punctuation
[ ] 7
8
8
tuv8TUV
8TUVtuv
‘ “ \ 8
9
9
wxyz9WXYZ
9WXYZwxyz
^ _ 9
0
0
@-_0=,<>
0-_
[space] 0
*
.*
.$*&%+()
*.
*.=+/-
#
@*
[space] , ( ) |
# [space]
#!?$%
Favorites menu
The Favorites menu assists you in dialing by providing access to a
predefined list of dial numbers. The predefined list can include either
complete dial numbers for named parties or partial numbers that
need additional data entry. This might be the case, for example, if a
PBX feature access code for call forwarding is defined in the favorites
list but you need to add the forwarding destination information
before sending the call to the PBX to activate the feature.
While in a dialing state, press the Favr softkey to display a menu of
pre-defined numbers or names that can be dialed (as programmed in
the SIP TFTP configuration file.) When an item is selected from the
list, the dial number is displayed. You may edit or add digits to the
displayed number if necessary before pressing the NavOK key or START
to place the call. When using the Favorites menu to perform a blind
transfer to someone on the list, select an entry from the Favorites
menu and then press FCN and select Transfer.
FCN menu
The FCN menu is accessible while in the active mode and provides
these features:
Transfer
Do Not Disturb
Set/Clear Forward
<OAI>
<OAI>
<OAI>
Items on this menu are accessible through navigation and selection
keys or through short-cut keys as displayed with the menu items.
OAI functions are automatically added as items at the end of this
menu when defined on an OAI server.
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Using the SIP Handset
Handset Operation
If you want to…
Then…
Turn the handset on Press and hold the END key until two chirps sound.
Turn the handset off Press and hold the END key. One chirp will sound. If you are in a call, hang up
first, then turn off the handset.
Unlock the keypad Press the Unlk softkey, then #.
Lock the keypad While in Standby press the Cfg softkey, then press NavOK.
Place a call
1 To dial a number, follow any one of these sequences:
»
»
»
»
Press the START key, wait for a dial tone, then dial the number. Then
press either START again or Nav OK.
Dial the number and then press the START key or Nav OK.
Press the Spkr softkey, then dial the number. Then press START or Nav
OK.
Press the START key; press the Favr softkey; use the Nav cd keys to
select the number or user from the list; press Nav OK to dial the
number.
2 Listen for the ring to indicate the alerting of the called party.
Note: Line 1 is the default line.
Place a call from
Favorites menu
1 Press START or Spkr.
2 Listen for dial tone.
3 Press the FAVR softkey.
4 Use the Nav keys to navigate to the desired entry.
5 Select the entry by pressing NavOK.
6 Press START or NavOK to place the call.
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
If you want to…
Place a second call
Then…
1 To get a dial tone for the second call, press LINE and NavOK.
a. If the handset has multiple lines, press LINE + [the line number].
b. Press NavOK.
• The first call is automatically placed on hold.
• The second call appears in a new call tab.
2 Dial the number to place the second call.
3 Use the Nav ef keys to toggle between calls.
Place a call on a
different line
1 Press the LINE key.
2 Navigate to the desired line and press NavOK or press the number of the
line.
3 Dial the number.
4 Press START or NavOK to initiate the call.
Answer a call
• Press START, and hold the handset to your ear.
• Press NAV OK, and hold the handset to your ear.
• Press the Answ softkey and hold the handset to your ear.
• Press the Spkr softkey and speak towards the handset.
Reject a call Press the Rej softkey to reject the call and allow the SIP server to redirect the
call elsewhere, usually to voicemail.
Answer a call on a If another call comes in on a different line, a new call icon flashes and a tone
second line sounds in the audio stream until the call is answered, the first call is terminated,
the caller hangs up, or the call transfers to voicemail.
1 To view the caller ID of the incoming call without interrupting the active call,
press Nav f. The original call’s audio remains active. The display now
shows information about the incoming call.
2 Press NavOK, START or Answ to place the current call on Hold and answer
the second call, or
Press Rej to redirect the incoming call to voicemail or as otherwise
programmed.
Navigate among call Use Nav► and Nav◄.
tabs
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Using the SIP Handset
If you want to…
Then…
Forward calls Calls may be forwarded on a per-line basis. The Line menu display will indicate
if a line is forwarded by a > character followed by the destination address.
• When a line is forwarded, the standby screen displays the forwarding
status, unless the do-not-disturb feature is enabled.
• If your handset has multiple lines, set or clear forwarding for each line
individually.
1 Press START or press Line and a line number to select the line that you want
to be forwarded.
2 Press FCN and select the Set/Clear Forward item from the FCN menu using
a digit key or NavOK key.
3 Enter the forwarding number for the indicated line.
4 Press NavOK to save.
To clear call forwarding for a given line #, again press START or select the line #.
Again select the Set/Clear Forward item from the menu.
Clear forwarding on a Press START or select the line number. Select the Set/Clear Forward item from
line the menu.
Listen to voicemail
1 Press START.
2 Press Msg softkey, or
Dial your voice message system number
3 Press START or Nav OK.
Do Not Disturb
1 Press START.
(reject all incoming
calls)
2 Press FCN.
3 Select the Do-Not-Disturb item from the menu and press NavOK.
When the do-not-disturb feature is enabled, the standby screen will display Do
Not Disturb. The handset will not ring for new incoming calls.
Clear Do Not Disturb
1 Press START.
2 Press FCN.
3 Select the Do-Not-Disturb item from the menu and press NavOK .
Redial the last
number you dialed
1 Press START.
2 Press the Redl softkey.
3 Verify that the displayed number is the one you want and the line number is
valid.
4 Press NavOK or START.
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
If you want to…
Activate installed
custom applications
(registered OAI
application on
an OAI server)
Then…
1 Press START or Spkr.
2 Press FCN.
3 Navigate to the desired custom application using Nav cd keys or the More
softkey.
4 Select the application using NavOK or shortcut key.
Transfer a call (blind)
1 While in a call, press FCN and then select the transfer item by shortcut key
or NavOK. (The current call will be placed on hold and it will be marked with
a transfer icon . A new call will be started and you will hear a dial tone.)
2 Dial the number to which you wish to transfer the call or press the Favr
softkey and select an entry from the Favorites menu.
3 Press FCN and select the Transfer option again. The marked call will be
transferred to the number that you have entered.
4 Press the END key to return to standby. If the transfer fails, you will see an
error message and you can then pick up the original call by navigating to
the marked call and pressing NavOK.
Transfer a call
(consulted)
1 While in a call, press FCN and then select the transfer item by shortcut key
or NavOK. (The current call will be placed on hold and it will be marked with
a transfer icon . A new call will be started and you will hear a dial tone.)
2 Dial the number to which you wish to transfer the call or press the Favr
softkey and select an entry from the Favorites menu.
3 Press NavOK or START to establish a second call.
4 Inform the person on the other end that you would like to transfer the call.
5 Press FCN and select the Transfer option to transfer the call.
6 Press the END key to return to standby.
Transfer an active call
to a call on Hold
1 Press FCN and then select the Transfer option by pressing NavOK. The
current call will be placed on hold and it will be marked with a transfer icon
.
2 Navigate to the second call (already on hold).
3 Press the Resm softkey and tell the other party that the call will be
transferred.
4 Press FCN and select the Transfer option by pressing NavOK.
5 Press the END key to return to standby.
Silence the ringing Press the END key to silence the external speaker ring and convert to in-ear
speaker ringing.
External speaker ringing will resume when the next incoming call is received
while the handset is in standby mode.
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Using the SIP Handset
If you want to…
Then…
Change the Press the up/down volume buttons on the side of the handset during ringing, or
ring volume while the handset is in standby mode.
Adjust the While speakerphone is active, press the up/down volume buttons on the side of
speaker volume the handset during the call.
Adjust the While the headset is plugged in, press the up/down volume buttons on the side
headset volume of the handset during the call.
Mute/Unmute a call Press the Mute softkey.
changes to the audio
When the handset is muted, the audio flowing icon
receive-only icon . Press the Mute softkey again to restore audio pickup.
End a call Press the End softkey to maintain the active mode and view the active calls.
Press the END key on the keypad to return to the standby mode.
Change the profile Press the Prof softkey and use the Nav keys to select a new profile while in
standby mode. The selected profile is marked with an asterisk (*).
Open Config menu Press the Cfg softkey from standby mode.
Turn on the backlight The backlight comes on when any key is pressed or when there is an incoming
call, and stays on for 10 seconds. It turns off if another key is not pressed within
that period.
Resume a call on hold Press the Resm softkey. If more than one call is on hold, use the Nav ef keys
from standby. to select the call you wish to resume and press the Resm softkey or NavOK.
Open the phonebook Press the Phbk softkey from standby mode. The phonebook may also be
opened by pressing the Save softkey when it appears in a call log or during
predialing.
The phonebook list is sorted alphabetically.
The <new> option appears until the maximum number of entries (20) has been
entered.
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
If you want to…
Then…
View system speed If the system speed dial key has been programmed, it will be listed as the first
dial number entry with a close bracket (>) as the first character. Only the system
administrator may change this entry.
Search for a Use NavS and NavT to scroll through the names or press the keys
phonebook entry corresponding to the first letters of the name. Use Nav◄ and Nav► to edit the
search characters as needed.
View a phonebook Select the name and press NavOK.
entry
The entry may be called, assigned a speed dial number or edited from this
screen. If a speed dial number has been assigned to this name, it will appear
beside spdl.
Return to the phonebook list by pressing the Back softkey or NavOK.
Edit a phonebook
entry
1 Select the name to edit from the phonebook list.
2 Press the Edit softkey to open the Edit Number display and edit the existing
number. Use the Clr and Del softkeys as needed.
3 Press NavOK to display the Edit Name display and edit the name.
4 Press NavOK to save the changes and exit the editing screens. Press the
Back softkey to exit without saving the changes.
Dial phonebook Select the entry and press START or the Spkr softkey.
number
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Using the SIP Handset
If you want to…
Then…
Enter a new name
and number in the
phonebook
1 Open the phonebook.
2 Select the <new> option (if available) and press the Edit softkey to open the
Enter Number display.
3 Enter the name and number by following the steps for editing a name and
number. You must enter alphanumeric characters for the name, not blank
spaces.
Delete a phonebook
entry
1 Open the phonebook and select the entry.
2 Press the Del softkey to delete the entry.
Open the speed dial Press the Spdl softkey from standby mode.
list from standby
Make a speed dial call
1 From standby or while in the speed dial menu, press and hold the
corresponding number key on the keypad for one second. The system
speed dial key must be pressed for three seconds.
2 The handset will display the name and number for one second before the
call is dialed. You may press END during this second to terminate the call.
(You may also use the Nav keys to highlight an entry and then press START
or the Spkr softkey to dial the number.)
If you do not place a call, you may return to standby by pressing the Exit softkey.
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
If you want to…
Then…
Assign a speed dial A phonebook entry may be assigned to a speed dial key from the speed dial list
number or from the phonebook edit number display.
From the speed dial list:
1 Select the speed dial key that you wish to assign.
2 Press the Set softkey to open the phonebook list.
3 Use the search routine or the Nav keys to select the desired entry.
4 Press NavOK to assign the selected name to that speed dial key.
5 Press NavOK again to exit to standby.
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Using the SIP Handset
If you want to…
Then…
From the phonebook view number display:
1 Press the Spdl softkey from the view number display in the phonebook.
2 When the speed dial list opens, navigate to an empty slot and press
NavOK. This sets the speed dial key to the number in the phonebook.
3 Press END to exit to standby mode and save the speed dial number
assignment.
Speed dial entries may be edited through the phonebook, as described above.
Push to Talk [the PTT Vibrate feature has been added]
PTT Vibrate To enhance the alerting of a PTT call, PTT Vibrate may be enabled. When PTT
Vibrate is enabled, the handset will vibrate three times whenever a PTT
broadcast is received, whether the handset is in standby or in a call. If in a call,
the chirp alert will also sound. The vibration does not replace any PTT tone
volume already set. To set the handset for PTT vibrate only, enable PTT Vibrate
and set the tone volume to zero.
PTT Vibrate is disabled by default.
Emergency Dial The large button on the left side of the SpectraLink 8030 Wireless Telephone
may provide you with either Emergency Dial or PTT functionality as determined
and activated by the system administrator.
To determine if Emergency Dial has been activated, press this button one time. If
Emergency Dial is active, an explanatory message will be displayed.
When Emergency Dial has been enabled, the handset will dial an emergency
number when it is pressed twice within two seconds. The number dialed may be
found under the Config Menu>System Info option.
Note that this feature is functional when the keypad is locked.
Caution! Emergency Dial will be inoperable if the wireless system or the call
server fails for any reason. Do not rely on it as the sole method of emergency
notification.
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7
Testing a Handset
Verify proper registration and operation of each handset by
performing the following tests on each handset in an active wireless
area.
1. Power on the handset by pressing the END key. A series of
messages will be displayed as the handset acquires the system.
The handset should display the user extension.
2. Place a call and listen to the audio quality. End the call by pressing
the END key.
3. Place a call to the handset and verify ring, answer, clear transmit,
and clear receive audio.
4. Use the softkeys to verify all softkey programmed features on the
handset.
5. Press the END key. Any line indicators should turn off and the
extension number display will return.
If any of these steps fails to operate as described, refer to Chapter 11
Troubleshooting for corrective action.
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101
8
Diagnostic Tools
Run Site Survey, Diagnostics Enabled and Syslog Mode are three diagnostic
tools provided to assist the LAN administrator in evaluating the
functioning of the SpectraLink 8020/8030 Wireless Telephone and the
system surrounding it. Diagnostic Tools are enabled in the Admin
menu.
The Halt on Error option in the Admin menu is a diagnostic tool that
will cause the handset to stop operating if an error message is
received. Error detail may be shown on the display, captured by the
syslog server, and may also be available for downloading with the
Handset Administration Tool. Unless the error is a fatal one, normal
operation may be resumed by power-cycling the handset.
Run Site Survey
Site survey is used to evaluate the facility coverage before certifying
that an installation is complete. It can also be used at any time to
evaluate coverage by testing signal strength, to gain information
about an AP, and to scan an area to look for all APs regardless of
SSID. The information available through the site survey includes:
•
SSID
•
Beacon Interval
•
AP information regarding support of 802.11d, 802.11h and other
802.11 amendment standards as required
•
Current security configuration
Start the site survey by selecting Run Site Survey from the Admin menu.
The mode starts immediately.
When the test is started, it is by default in “single SSID” mode. When
the Any soft key is pressed (softkey A) all APs, regardless of SSID, are
displayed and the softkey changes to say MyID. Pressing the MyID soft
key will revert to the “single SSID” mode and change the softkey back
to Any.
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
The display would look like the following for the single AP mode.
1 1 1 1 1 1
- 2 2
3 3
4 4 4
1 1 1 1 1 1
- 2 2
3 3
4 4 4
1 1 1 1 1 1
- 2 2
3 3
4 4 4
1 1 1 1 1 1
- 2 2
3 3
4 4 4
A n y
D e t l
Where:
•
111111 – the last three octets of the on-air MAC address for a
discovered AP.
•
22 – the signal strength for the specified AP.
•
33 – the channel number of the specified AP.
•
444 – the beacon interval configured on the specified AP.
•
Any/MyID – softkey to toggle between “single SSID” and “any
SSID” mode.
•
Detl/Smry – softkey to toggle between the multiple AP
(summary) display, and the single (detail) displays for each AP.
The following screen shows how the display would look when there
are three APs configured with an SSID that matches that of the
handset. The first has a signal strength of –28 dBm, is configured on
channel 2, with a beacon interval of 100 ms. The second has a signal
strength of –48 dBm, is configured on channel 6, with a beacon
interval of 200 ms. The third has a signal strength of –56 dBm, is
configured on channel 11 with a beacon interval of 100 ms.
a b 7 b c 8
- 2 8
0 2
1 0 0
2 a e 5 7 8
- 4 8
0 6
2 0 0
2 a e 5 9 6
- 5 6
1 1
1 0 0
A n y
104
D e t l
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Diagnostic Tools
When the Any SSID mode is selected, the summary display contains
the first six characters of the APs SSID instead of the beacon interval
as in the example below.
a b 7 b
- 2 8
0 2
A L P H A
2 a e 5
- 4 8
0 6
W S M T E S
2 a e 5
- 5 6
1 1
v
o i
M y I D
c e
D e t l
In Detl (detail) mode the display would appear as follows. The
left/right arrow keys will move between AP indices.
i
:
b b b b b b
s n
e e e e e e e e e e e
r
c h
b c n
D G H I
r r r r r r r r r r r r + x x x x
Q : X P
C : v C
s s s s s s s
A n y
S m r y
Where:
•
i – index of selected AP (value will be from 0 to 3 inclusive)
•
bbbbbb – the last three octets of the BSSID for a discovered AP
•
sn – signal strength in –dBm
•
ch – channel
•
bcn – beacon interval
•
eeeeeeeeeee – SSID (up to first 11 characters)
•
DGHI – standards supported i.e. 802.11d, 802.11g, etc. in addition
to 802.11a and 802.11b.
•
rrrrrrrr – rates supported. Basic rates will have a “b” following the
rate
•
+ – more rates are supported than those displayed
•
xxxx – WMM or UPSD if those QoS methods are supported
•
Q:XP
o
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X is a hexadecimal representation of the access categories
configured with admission control mandatory (ACM).
Bit3 = voice, Bit2 = video, Bit1 = background, Bit0 = best
effort. For example, if an AP advertises voice and video as
ACM then X=c. If all the ACs are set as ACM then X=f. If
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
AP does not have WMM support, this character space will
be blank.
o
•
P is displayed when the AP advertises WMM-PS. If the AP
does not advertise WMM-PS then this character space will
be blank.
C:vC
o
v is a decimal number indicating the CCX version
advertised by the AP.
o
C is displayed when AP advertises CCKM. If the AP does
not advertise CCKM then this character space will be
blank.
•
ssssssss – Security modes: “None”, “WEP”, “WPA-PSK”, “WPA2PSK”, “WPA2-Ent”
•
Any/MyID – softkey to toggle between “single SSID” and “any
SSID” modes
•
Detl/Smry – softkey to toggle between the multiple AP display
(summary), and the single AP display (detail)
Numbers racing across the handset display indicate AP information is
being obtained. A Waiting message indicates the system is not
configured properly and the handset cannot find any APs.
Solving coverage issues
Coverage issues are best resolved by adding and/or relocating APs.
Overlap issues may be resolved by reassigning channels to the APs or
by relocating them. See Chapter 11 Troubleshooting, section Access
Point Problems for more information.
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Diagnostic Tools
Diagnostics Enabled
Diagnostics is used to evaluate the overall quality of the link between
the handset, AP, and infrastructure side equipment, such as IP PBX,
SpectraLink 8000 SVP Server, and gateways. Unlike Site Survey,
Diagnostics is used while the functional code is running, and during a
call.
When Diagnostics is enabled in the Admin menu, the handset can
display diagnostic screens any time it is in active mode.
The display of information is instigated, when in call, by pressing the
Nav◄ or Nav► key. Only one of the six diagnostic screens listed
below can be shown at a time. Pressing the Nav keys multiple times
will cycle through the various diagnostics screens and the normal offhook (IP-PBX) display. The numeric icon at the top of the display
indicates what screen number is being displayed. For example: The
first time the Nav key is pressed, the 1 icon is shown, and the first of
six diagnostics screens are displayed. The next time it is pressed, the 2
icon is shown, and the next of six diagnostics screens are displayed.
The counters will be cycled through in this fashion until there are no
more counters to be displayed. After all the diagnostics screens have
been displayed, the screen returns to the normal off-hook IP-PBX
screen.
The information provided by Diagnostics includes:
Screen 1
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•
Missed receive packet count since power up (MissedRcvCnt)
•
Missed transmit packet count since power up (MissedXmtCnt)
•
Receive retry count since power up (RxRetryCount)
•
Transmit retry count since power up (TxRetryCount)
M i s s e d R c V C n t
n n n n n
M i s s e d X m T C n t
n n n n n
R x R e t r y C O u n t
n n n n n
T x R e t r y C O u n t
n n n n n
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SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Screen 2
•
Jitter – average error or “wobble” in received packet timing, in
microseconds
•
Last successful transmit data rate (LastRate)
•
Gateway type (GatewyType)
•
TX Power (dBm)
J i
t t e r
n n n n n
L a s t R a t e
m m m m m
G a t e w y T y P e
m n e m o
T X P o w e r ( D B m )
r
r
r
r
r
Where:
•
nnnnn – Jitter in micro-seconds
•
mmmmm – last transmit rate
— 11Mb – SpectraLink System with SVP Server. Handsets can
run at full 11Mb when in 802.11b/g mode
— 54Mb – SpectraLink System with SVP Server. Handsets can
run at full 54Mb when in 802.11a or 802/11g modes
•
mnemo – a mnemonic that indicates what type of gateway is
being used
•
rrrrr – TX Power configured in dBm
Screen 3
Screen 3 contains a list of the APs that are heard and the following
parameters from each AP:
108
•
Indicator as to whether this is the current AP or an index into the
list of other APs heard (C indicates current)
•
Last 2 octets of the MAC address of the AP (mmmm)
•
Channel number (ch)
•
Signal strength (ss)
•
Either the 802.11 Association ID from the current AP or a
mnemonic for the reason code indicating why the handset didn’t
hand off to this other AP
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Diagnostic Tools
m m m m
c h
- s s
a i d
m m m m
c h
- s s
m n e m
m m m m
c h
- s s
m n e m
m m m m
c h
- s s
m n e m
Where:
•
mmmm – This hexadecimal number is the last 2 octets of this AP’s
MAC address
•
ch – Channel number the AP is configured on
•
-ss – Signal strength for the AP in dBm
•
aid – The Association ID for the currently associated AP
AP mnem – a mnemonic indicating the reason code:
•
Unkn – reason unknown
•
Weak – signal strength too weak
•
Rate – one or more basic rates not supported
•
Full – AP can not handle bandwidth requirements
•
AthT – authentication timeout
•
AscT – association timeout
•
AthF – authentication failure
•
AscF – association failure
•
SecT – security handshake timeout
•
SecF – security handshake failure
•
Cnfg – AP not configured correctly for security, QoS mode or
infrastructure network
•
CCX – AP is not CCX complaint.
•
CCKM – AP does not support CCKM.
•
WMM – AP does not meet the WMM requirements. Probable
reason could be that the access categories AC_VO and AC_VI,
used for voice and control traffic respectively, might not be
marked as mandatory or WMM-PS might not be enabled.
Screen 4
•
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Association count since power up (AssocCount)
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•
Re-association count since power up (ReAssocCount)
•
Association failures since power up (AssocFailure)
•
Re-association failures since power up (ReAssocFail)
A s s o c C o u n t
n n n n n
R e A s s o c C o u n t
n n n n n
A s s o c F a i
n n n n n
l u r e
R e A s s o c F a i l
n n n n n
Screen 5
•
WEP only security error count since power up (Sec-ErrCount)
•
MAC sequence number of frame with last security error
(LstSecErrSeq)
•
(Re)Association failures due to QoS (QoSFailCnt). Usually
attributed to insufficient available bandwidth on an AP.
S e c - E r r C o u n t
L s t S e c E r r S e q
Q o S F a i l C n t
n n n n n
n n n n n
n n n n n
Screen 6 – EAP Information
110
•
“xxxxx” in Line 1 is a 5-digit decimal value displaying the EAP
authentication failure/error count.
•
“xxxxx” in Line 2 is a 5-digit decimal value displaying the error
code/sequence for the last EAP authentication reason, listed just
below. Line 2 will be blank if the count for Line 1 is zero.
•
1 = Unknown error
•
2 = Mismatch in EAP type. The phone is configured with an EAP
type (Cisco FSR, PEAP or EAP-FAST) that is not supported by the
AP.
•
3xxx = Certification failure. The certificate presented by the server
is found as invalid. “xxx” when having a non-zero value, is the
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Diagnostic Tools
standard TLS alert message code. For example, if a bad/invalid
certificate (on the basis of its signature and/or content) is
presented by the server “xxx” will be 042. If the exact reason for
the certificate being invalid is not known, then the generic
certificate error code would be xxx=000. [Refer
http://www.ietf.org/rfc/rfc2246.txt , section 7.2 for further TLS
alert/error codes].
•
4xxx = Other TLS failures. This is due to TLS failure other than
certification related errors. The reason code (the TLS alert
message code) is represented by “xxx”. For example, if the
protocol version presented by the server is not supported by the
phone then xxx will be 70, and the EAP error code would be 4070.
[Refer http://www.ietf.org/rfc/rfc2246.txt , section 7.2 for further
TLS alert/error codes].
•
5xxx = Security credential Failure. This is due to an invalid
username and/or password produced by the phone. xxx when
non-zero, presented the 3-digit error code sent by the server in
response to phone’s credential. For example, if the server has sent
the error code as “691”, then the EAP error code would be 5691. If
the server does not send the error code message, then xxx is
defaulted to 000, i.e., EAP error code would be 5000. Refer [1])
http://www.ietf.org/rfc/rfc2759.txt section 6, [2]
http://ietfreport.isoc.org/all-ids/draft-zhou-emu-fast-gtc-02.txt
section 2.
E A P E r r C n t
x x x x x
L a s t E A P E r C o d e x x x x x
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Syslog Mode
A syslog server must be present on the network in order for the
handset to send the log messages and have them saved. The syslog
server will be found with DHCP option 7 if the handset is using
DHCP. If static addresses are configured, the syslog server’s IP
address can be configured statically in the Admin menu.
If the syslog server address is blank (000.000.000.000 or
255.255.255.255) or the handset is using DHCP and no option 7 is
received from the DHCP server, the handset will not send any
syslog messages.
Admin menu options:
•
*Disabled – turns syslog off.
•
Errors – causes the handset to log only events that we consider to
be an error (see below).
•
Events – logs all errors plus some other interesting events (see
below).
•
Full – logs all the above plus a running stream of other quality
information (see below).
The table below lists the syslog messages and which level of logging
will produce them:
112
Message type
Failed Handoff
Errors
Yes
Events
Yes
Full
Yes
Successful Handoff
No
Yes
Yes
Security Error
Yes
Yes
Yes
Call Start/End
No
Yes
Yes
Audio stats
No
No
Yes (every 5 secs)
Audio error threshold exceeded
Yes
Yes
Yes
Radio stats
No
No
Yes (every 5 secs)
Radio error threshold exceeded
Yes
Yes
Yes
Error Handling Mode
Yes
Yes
Yes
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All syslog messages will include:
•
Date and time (to 1/100th of second) since handset power on (The
handset time is set when it is powered on to Jan-1 00:00.00 GMT
adjusted. If it has obtained a time from the network time server,
that time will display instead.)
•
Handset’s MAC address
•
Handset’s IP address
•
Sequence number
The table below lists the additional items in each message type:
Failed Handoff
Failed AP MAC
(Sent whenever the handset
attempted to handoff, but failed
trying.)
Failed AP signal strength
Current AP MAC
Current AP signal strength
Failure reason
Wireless Telephone Transmit Power to Old AP
Wireless Telephone Transmit Power to New AP
FCCKM‡ – Failed to use CCKM for fast handoff
FOKC‡ – Failed to use OKC for fast handoff
Successful Handoff
New AP MAC
New AP signal strength
Old AP MAC
Old AP signal strength
Reason for handoff
Other candidate APs:
MAC
Signal strength
Reason not used
Wireless Telephone Transmit Power to Old AP
Wireless Telephone Transmit Power to New AP
FCCKM‡ – Failed to use CCKM for fast handoff
FOKC‡ – Failed to use OKC for fast handoff
Security Error
AP MAC
AP signal strength
Security mode
Error details (mode-dependent)
Call Start
Call type (telephony, OAI, PTT)
AP MAC
AP signal strength
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Call End
AP MAC
AP signal strength
Audio stats
AP MAC
AP signal strength
Payload size (in msec)
Payloads sent
Payloads received
Payloads missed (not received)
Payloads missed rate (over last 5 seconds)
Payloads late
Payloads late rate (over last 5 seconds)
Average jitter
Audio error threshold exceeded
Same as audio stats
(Sent if payloads missed rate or
payloads late rate exceeds 2%, or
if the average jitter is over 2 msec)
Radio stats
AP MAC
AP signal strength
Directed packets sent
Directed packets received
Multicast packets sent
Multicast packets received
Broadcast packets sent
Broadcast packets received
TX dropped count
TX drop rate (over last 5 seconds)
TX retry count
TX retry rate (over last 5 seconds)
RX retry count
RX retry rate (over last 5 seconds)
Radio error threshold exceeded
Same as radio stats
(Sent if TX drop rate exceeds 2%
or TX or RX retry rate exceeds
5%)
Probe Recovery
Probe Recovery Count
LockUpRecovery
Lockup Recovery Count
DCA initiated radio reset
Reset count when Reset occurred
Reset count at the time when the syslog was sent
‡ Present only when the specific fast handoff method (CCKM, OKC)
has been enabled.
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Diagnostic Tools
Messages are formatted like the following example:
Jan 1 00:01:26.72 0090.7a02.2a1b (172.16.0.46) [001a]
RStat: AP 00:40:96:48:1D:0C (-56 dBm), Sent 783523,
Recvd 791342, MSnt 245, MRcd 5674, BSnt 43, BRcd 10783,
TX drop 43 (0.0%), TX retry 578 (1.2%), RX retry 1217
(1.6%)
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9
Certifying the Handsets
Prior to determining that an installation is complete, test the handsets
following the sequence given in the previous Testing a Handset section
and conduct a Site Survey mode test according to the directions given
in the previous Diagnostic Tools section.
The installation may need some adjustments. Note any areas where
coverage is conflicting or inadequate. Note any system difficulties and
work with your wireless LAN and/or LAN system administrator to
determine the cause and possible remedy. See Chapter 11
Troubleshooting for clues to possible sources of difficulties. If any
adjustments are made to the system, re-test the device in the same
vicinity to determine if the difficulty is resolved.
The installer should not leave the site before performing installation
verification.
These tests must be performed in typical operating conditions,
especially if heavy loads occur. Testing sequence and procedure is
different for every installation. Generally, you should organize the
test according to area and volume, placing numerous calls to others
who can listen while you perform coverage tests. Note any areas with
excessive static or clarity problems and report it to a Polycom service
engineer.
The coverage test will also require you to put the handset in Site Survey
mode and walk the entire coverage area to verify all APs.
Conducting a Site Survey
Conduct a site survey of the installation, by walking the site looking
for interfering 802.11 systems, adequate coverage and channel
assignment, and correct AP configuration. The site survey discussed
here does not replace an RF site survey conducted by professionals
who specialize in WLAN design and voice optimization
implementations. Polycom offers professional services including RF
site surveys.
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The handset’s site survey mode is not a replacement for a
professional site analysis and should be used only for testing,
limited site validation, and troubleshooting.
The handset’s site survey mode does not include functionality to
allow for analysis or troubleshooting of 802.11n specific WLAN
features.
1. Referring to Chapter 8 Diagnostic Tools, section Run Site Survey,
put a handset into Site Survey in the Any/Smry ESSID mode. Walk
throughout the site checking for any expected APs or other
ESSIDs.
2. Then, walk the site again, in MyID/Smry ESSID mode, this time
checking that every location has adequate coverage and has good
channel allocation.
There should be at least one AP stronger than -the minimum
specified in the following tables.
At any point, the strongest AP shown should be on a different
channel than the next best choice.
The handset configured for 802.11b requires:
— -70dBm when all 802.11b data rates are available (with only
1Mbps set Required)
— -65dBm when only 2Mbps is set Required and other higher
rates enabled
— -64dBm when only 5.5Mbps is set Required with 11Mbps set
enabled
— -60dBm when 11Mbps is set required and other 802.11b rates
disable or enabled
802.11
Radio Standard
802.11b
-70 dBm
Maximum
“Mandatory” Data
Rate
1 Mb/s
-60 dBm
11 Mb/s
Minimum Available
Signal Strength (RSSI)
— The critical factor is the highest data rate set Required or
Mandatory. Other 802.11b data rates can be set enabled or
disabled. The highest data rate set Required or Mandatory
determines the RF power available to the wireless telephone
for proper operation.
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Certifying the Handsets
The handset configured for 802.11g requires:
— -60dBm when all 802.11g data are available (with only 6Mbps
set Required)
— -45dBm when 54Mbps is set Required and other 802.11g rates
Required, Enabled or Disable
802.11
Radio Standard
802.11g
-60 dBm
Maximum
“Mandatory” Data
Rate
6 Mb/s
-45 dBm
54 Mb/s
Minimum Available
Signal Strength (RSSI)
— The critical factor is the highest data rate set Required or
Mandatory. Other 802.11g data rates can be set Required,
Enabled or Disabled. The highest data rate set Required or
Mandatory determines the RF power available to the wireless
telephone for proper operation.
— -45dBm when 54Mbps is set Required and other 802.11g rates
Required, Enabled or Disable
The handset configured for 802.11g requires:
— -60dBm when all 802.11a data are available (with only 6Mbps
set Required)
— -45dBm when 54bps is set Required and other data rates
Required, Enabled or Disabled
802.11
Radio Standard
802.11a
-60 dBm
Maximum
“Mandatory” Data
Rate
6 Mb/s
-45 dBm
54 Mb/s
Minimum Available
Signal Strength (RSSI)
— The critical factor is the highest data rate set Required or
Mandatory. Other 802.11a data rates can be set enabled or
disabled. The highest data rate set Required or Mandatory
determines the RF power available to the wireless telephone
for proper operation.
3. Finally, use the single AP (MyID/Detl) display to check each AP, to
ensure it is configured for the proper data rates, beacon interval,
802.11 options enabled, QoS method, and security method.
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Make any necessary adjustments to AP locations and configurations
and repeat steps 1 through 3 until the site survey shows adequate
coverage and correct configuration at every location.
The installation is not complete until these certification steps have
been performed. Do not hand out handsets at a site that has not
been certified.
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10
Software Maintenance
The SpectraLink 8020/8030 Wireless Telephones use proprietary
software programs written and maintained by Polycom Corporation.
The software versions that are running on the handsets can be
displayed during power on by holding down the END button. Firmware
Version is also an option on the Config menu.
Polycom Customer Service or an authorized dealer will provide
information about software updates and how to obtain the software
(for example, downloading from a website).
After software updates are obtained, they must be transferred to the
appropriate TFTP server located on the LAN to update the code used
by the Wireless Telephone.
The handset allows over-the-air transfer of software updates from the
designated TFTP server to the handsets. The download function in the
Wireless Telephone checks its software version every time the
handset is powered on, when the TFTP server is active. If there is a
different version available, the handset immediately begins to
download the update.
Upgrading Handsets
After software updates are obtained from Polycom, they must be
transferred to the appropriate location in the LAN to update the code
used by the handsets.
SpectraLink 8020/8030 Wireless Telephones allow over-the-air
transfer of software updates from the designated TFTP server to the
handsets. The downloader function in the handset checks its software
version every time the handset is turned on. If there is any
discrepancy the handset immediately begins to download the update.
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Normal Download Messages
When the handset is powered on, it displays a series of messages
indicating that it is searching for new software, checking the versions,
and downloading. The normal message progression is:
Message
Checking Code
Description
Handset is contacting the TFTP server to determine if it has a newer version of
software that should be downloaded.
Erasing Memory
Handset has determined that a download should occur and is erasing the current
software from memory. This message also displays a progress bar. When the
progress bar fills the display line the erase operation is complete.
Updating Code
Handset is downloading new software into memory. The number icons at the
bottom of the display indicate which file number is currently being downloaded.
This message also displays a progress bar. When the progress bar fills the display
line the update operation is complete on that file.
When the update is complete, the handset displays the extension
number, and is ready for use.
Download Failure or Recovery Messages
The following display messages indicate a failure or recovery
situation during the download process.
Message
Server Busy
Description
Handset is attempting to download from a TFTP server that is busy downloading
other phones and refusing additional downloads. The handset will automatically
retry the download every few seconds.
TFTP
ERROR(x):yy
A failure has occurred during the TFTP download of one of the files. (x) = The file
number which was being downloaded; yy is an error code describing the particular
failure. Possible error codes are:
01 = TFTP server did not find the requested file.
02 = Access violation (reported from TFTP server).
07 = TFTP server reported "No such user" error. Check the TFTP server
configuration.
16 = No TFTP server address. Check the TFTP server configuration.
81 = File put into memory did not CRC. The handset will attempt to download the
file again.
FF = Timeout error. TFTP server did not respond within a specified period of time.
Erase Failed
Download process failed to erase the memory in the handset. This operation will
retry.
Waiting
Handset has attempted some operation several times and failed, and is now
waiting for a period of time before attempting that operation again.
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11
Troubleshooting
On occasion, you may run into transmission problems due to any
number of factors originating from the wireless LAN. SpectraLink
8020/8030 Wireless Telephones can exhibit transmission problems in
several ways. They can cease functioning properly, display error
messages, or display incorrect data. When using and troubleshooting
handsets, consider the following problem sources to determine the
best method of approaching any specific situation.
Access Point Problems
Most, but not all, handset audio problems have to do with AP range,
positioning, and capacity. Performing a site survey as described in
this document can isolate the AP causing these types of problems. If
the handset itself is suspected, conduct a parallel site survey with a
handset that is known to be properly functioning.
In range/out-of-range
Service will be disrupted if a user moves outside the area covered by
the wireless LAN APs. Service is restored if the user moves back
within range. If a call drops because a user moves out-of-range, the
handset will recover the call if the user moves back into range within
a few seconds.
Capacity
In areas of heavy use, the call capacity of a particular AP may be
filled. If this happens, the user will hear three chirps from the
handset. The user can wait until another user terminates a call or
move within range of another AP and try the call again. If a user is on
a call and moves into an area where capacity is full, the system
attempts to find another AP. Due to range limitations, this may be the
same as moving out of range.
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Transmission obstructions
Prior to system installation, the best location for APs for optimum
transmission coverage should have been determined. However, small
pockets of obstruction may still be present, or obstructions may be
introduced into the facility after system installation. This loss of
service can be restored by moving out of the obstructed area or by
adding/rearranging APs.
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Handset Status Messages
SpectraLink 8020/8030 Wireless Telephone status messages provide
information about the SpectraLink 8020/8030 Wireless Telephone’s
communication with the AP and host telephone system. The
following table summarizes, in alphabetical order, the status
messages.
Message
Description
Download failure icon
Action
Update handset code in the TFTP
server and power cycle the handset.
3 chirps (audio)
Handset is not able to communicate
with the best AP, probably because
that AP has no bandwidth available.
None. This is only a warning, the call
will hand off to the best AP once it
becomes available.
802.1X Failure
When WPA2-Enterprise or Cisco FSR is
selected, the handset failed to connect
because the user credentials are restricted
based on the user account properties. In the
case of EAP-FAST, the PAC ID may not match
the username.
Verify and resolve if the user account
has any restrictions such password
expired, account restricted/ disabled, or
in case of EAP-FAST, the handset PAC
and username matching the
authentication server.
Xxxxxxxxxxxx
XXX
The second line of the error message
contains the twelve digits of the AP
MAC address and three digits that
indicate the error code as defined in
RFC2759.
Address Mismatch
Handset software download files are
incorrect or corrupted.
Download new software from the
Polycom website per Software
Maintenance.
Assoc Failed
xxxxxxxxxxxx
x…x = AP MAC address.
Handset association was refused by
AP; displays MAC of failing AP.
Check handset and AP security
settings.
Ensure AP is configured per VIEW
Configuration Guide.
Try another AP.
Assoc Timeout
xxxxxxxxxxxx
x…x = AP MAC address.
Handset did not receive association
response from AP; displays MAC of
failing AP.
Check handset and AP security
settings.
Ensure AP is configured per VIEW
Configuration Guide.
Try another AP.
Auth Failed
xxxxxxxxxxxx
x…x = AP MAC address.
Handset authentication was refused by
AP; displays MAC of failing AP.
Check handset and AP security
settings.
Ensure AP is configured per VIEW
Configuration Guide.
Try another AP.
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Message
Auth Timeout
xxxxxxxxxxxx
Description
x…x = AP MAC address.
Handset did not receive authentication
response from AP; displays MAC of
failing AP.
Action
Check handset and AP security
settings.
Ensure AP is configured per VIEW
Configuration Guide.
Try another AP.
Bad Code Type xx
Expected Code
Type yy
xx, yy = software license types.
Handset software does not match
current handset license selection.
Download new software from the
Polycom website per Software
Maintenance.
Bad Config
Some needed configuration parameter
has not been set.
Check all required handset
configuration parameters for valid
settings.
Bad SSID
The handset has not had an SSID
entered.
Statically configure an SSID in the
Admin menu.
Bad Phintl File
Handset software download files are
incorrect or corrupted.
Download new software from the
Polycom website per Software
Maintenance.
Bad Program File
Handset software download files are
incorrect or corrupted.
Download new software from the
Polycom website per Software
Maintenance.
Bad SIP TFTP IP
A bad unicast address has been
entered for the SIP TFTP server in
static entry mode.
Re-enter the correct IP address in the
administrative menus for static IP
addresses.
(battery icon),
Battery Low,
beep (audio)
Low battery.
In call: the battery icon displays and a
soft beep will be heard when the user is
on the handset and the battery charge
is low. User has 15–30 minutes of
battery life left.
Not in call: The battery icon displays
whenever the battery charge is low The
message Battery Low and a beep
indicate a critically low battery charge
when user is not on the handset. The
handset will not work until the Battery
Pack is charged.
Battery Failure
The Battery Pack is not functioning.
Replace the Battery Pack with a new or
confirmed SpectraLink Battery Pack.
Only SpectraLink Battery Packs will
work.
Battery Failed
Battery Pack is damaged or
incompatible with handset.
Replace the Battery Pack with a new or
confirmed SpectraLink Battery Pack.
Only SpectraLink Battery Packs will
work.
Can’t Renew
DHCP
yyy.yyy.yyy.yyy
y…y = DHCP server IP address.
DHCP server is not responding to
initial renewal attempt.
Configuration problem. Check the IP
address configuration in the DHCP
server.
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Message
Cert Expired
Description
When WPA2-Enterprise with PEAP
authentication is selected, the handset failed to
connect due to an expired certificate on the
handset or authentication server.
Action
Verify that the NTP server is properly configured
with the correct time.
Verify that the certificates loaded on the handset
and authentication server have valid start/end
dates by looking at “valid to” field from “validity”
data in certificates.
If any of the certificates have expired
replace them with new certificates.
Cert Invalid
When WPA2-Enterprise with PEAP
authentication is selected, the Wireless
Telephone failed to connect to the network
because the certificate start date is in the
future.
Verify that the NTP server is properly configured
with the correct time.
Verify that the certificates loaded on the handset
and authentication server have valid start/end
dates by looking at “valid from” field from
“validity” data in certificates.
If any of the certificates have expired
replace them with new certificates.
Charging …
The handset is charging in the desktop
charger.
No action needed.
Charge Complete
The handset is now fully charged.
No action needed.
Charger Error
The handset has detected a problem
with the charging circuitry.
Allow the charger and battery to cool. If
the problem persists, try a new or
confirmed battery. If the problem still
persists, contact technical support and
report the error.
Checking Code
Handset is contacting the TFTP server
to determine if it has a newer version
of software that should be
downloaded.
None, this message should only last for
approximately one second. If message
remains displayed, power off and
contact customer support for a
replacement phone.
Checking DHCP
IP
The handset is retrieving DHCP
information from the DHCP server.
None. This is informational only.
CRC Code Error
The software which has been TFTP
downloaded has a bad redundancy
code check.
Try the download again; it is possible
the software was corrupted during
download. If the error repeats, check
that the download image on the TFTP
server is not corrupted.
Code Mismatch!
The software loaded into the handset
is incorrect for this model handset.
Verify the License Management value is
correct. Replace the software image on
the TFTP server with software that is
correct for the handset model.
DCA Timeout
The handset has detected a fault for
which it cannot recover, possibly due
to a failure to acquire any network.
Turn the handset off, then on again. If
error persists, contact Polycom
Technical Support and report the error.
DHCP Error (1-5)
DHCP Error 1.
The handset cannot locate a DHCP
server. It will try every four seconds until
a server is located.
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Message
Description
DHCP Error 2.
Action
The handset has not received a
response from the server for a request
to an IP address. It will retry until a
server is found.
DHCP Error 3.
The server refuses to lease the handset
an IP address. It will keep trying.
DHCP Error 4.
The server offered the handset a lease
that is too short. The minimum lease
time is 10 minutes but Polycom
Engineers recommend at least onehour minimum lease time. The handset
will stop trying. Reconfigure the server
and power cycle the handset.
DHCP Error 5.
Failure during WEP Key rotation
process (proprietary feature).
DHCP Lease Exp
yyy.yyy.yyy.yyy
y…y = DHCP server IP address.
The handset failed to renew its DHCP
lease, either because the DHCP server
is not running, or because the
configuration has been changed by the
administrator. The handset will attempt
to negotiate a new lease, which will
either work, or it will change to one of
the above DHCP errors (1 through 4).
DHCP NACK error
yyy.yyy.yyy.yyy
y…y = DHCP server IP address.
DHCP server explicitly refused
renewal.
The DHCP lease currently in use by the
handset is no longer valid, which forces
the handset to restart. This problem
should resolve itself on the restart. If it
does not, the problem is in the DHCP
server.
DL Not On Sector
Handset software download files are
incorrect or corrupted.
Download new software from the
Polycom website per Software
Maintenance.
DO NOT POWER
OFF
The handset is in a critical section of
the software update.
None. Do not remove the Battery Pack
or attempt to power off the phone while
this is displayed. Doing so may require
the handset inoperable.
Duplicate IP
The handset has detected another
device with its same IP address.
If using DHCP, check that the DHCP
server is properly configured to avoid
duplicate addresses.
DHCP is not responding to renewal
attempts (at least one renewal
succeeded).
If using Static IP, check that the
handset was assigned a unique
address.
Erase Failed
128
Download process failed to erase the
memory in the handset.
Operation will retry but may eventually
report the error "int. error: 0F" Power
cycle the handset.
PN: 1725-36038-001-H.doc
Troubleshooting
Message
Erasing Memory
Description
Handset has determined that a
download should occur and is erasing
the current software from memory.
Action
None. When the progress bar fills the
display line the erase operation is
complete.
Do not turn the handset off during this
operation.
Files Too Big
Handset software download files are
incorrect or corrupted.
Download new software from the
Polycom website per Software
Maintenance.
Flash Config Error
Handset internal configuration is
corrupt.
Perform "Restore Defaults" operation
via administrator menus (or re-program
with Configuration Cradle).
Initializing …
The handset is performing power-on
initialization.
None. This is informational only.
Initializing SIP
The handset is performing a power-on
initialization of the SIP application.
The phone is initializing its data
structures and attempting to access
the SIP TFTP server and download the
SIP configuration files.
None. This is informational only.
Internal Err. # #
The handset has detected a fault from
which it cannot recover.
Record the error code so it can be
reported.
Turn the handset off then on again.
If error persists, try registering a
different handset to this telephone port.
If error still persists, contact Polycom
Technical Support and report the error.
Invalid Usr/Pwd
When WPA2-Enterprise or Cisco FSR
is selected, the handset failed to
connect due to incorrect device
credentials or unavailability of
authentication server. If the error is
because of the incorrect device
credentials then the username or
password doesn’t match with those
configured on the authentication
server.
Verify that the required credentials
{username, password} are created on
the authentication server and should
match the handset. This may also
happen when the authentication server
is not reachable while doing the EAP
authentication. Make sure the
authentication server is active and
reachable from the WLAN access
points/controller at all times.
Multiple GW Res
More than one SpectraLink 8000 SVP
Server has responded.
Caused by two or more handsets
sharing the same IP address. Assign
unique IP addresses to each handset.
Multiple SVP Reg
yyy.yyy.yyy.yyy
y…y = SVP IP address
Handset received responses from
multiple SVP Servers; displays IP
address of one responding SVP
Server.
This can happen if the handset has
been reconfigured to use a different
SVP server and then powered on
before the previous server has had time
to determine that the handset is no
longer connected to it. The problem
should go away after about 30 seconds.
PN: 1725-36038-001-H.doc
129
SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Message
Must Upgrade
SW!
Description
Handset software is incompatible with
hardware.
Action
Download new software from the
Polycom website per Software
Maintenance.
Net Busy
xxxxxxxxxxxx
x…x = AP MAC address.
Handset cannot obtain sufficient
bandwidth to support a call; displays
MAC of failing AP.
Try the call again later.
No 802.11a Subbands Enabled
'a' radio selected but no sub-bands are
enabled
Configure 'a' radio sub-bands from
Admin menus
No 802.11 Subbands Enabled
'b/g radio selected but no sub-bands
are enabled
Configure 'b/g' radio sub-bands from
Admin menus
No APs Heard
The handset is unable to hear
beacons/probes from any AP in the
network in site survey mode.
Verify that network is properly configured and the
handset is able to hear beacons from the AP.
No DHCP Server
Handset is unable to contact the
DHCP server.
Check that DHCP is operational and
connected to WLAN or use Static IP
configuration in the handset.
No ESSID
Attempted to run Site Survey
application without an ESSID set.
Let handset come completely up.
Statically configure an ESSID in the
Admin menu.
No Func Code
Handset software download files are
incorrect or corrupted.
Reconfigure the handset to gain access
to the WLAN and download new code.
No Host IP
The handset is configured for “static
IP” (as opposed to “use DHCP”) and
no valid host IP address (the handset’s
IP address) has been entered.
Enter a valid IP address in the
configuration settings or change to “use
DHCP.”
No IP Address
Invalid IP.
Check the IP address of the handset
and reconfigure if required.
No Net Access
Cannot authenticate / associate with
AP.
Verify the AP configuration.
No Net Found
No APs
130
Verify that all the WEP settings in the
handset match those in the APs.
This indicates that the handset cannot
find any access points and has no
additional information to display as to
why. Possible problems are
enumerated below.
No radio link.
Verify that the AP is turned on.
No ESSID: Auto-learn not supported
(or) incorrect ESSID.
Verify the ESSID of the wireless LAN
and enter or Autolearn it again if
required.
AP does not support appropriate data
rates.
Check the AP configuration against
Configuration Guide for AP.
Out of range.
Try getting closer to an AP. Check to
see if other handsets are working within
the same range of an AP. If so, check
the ESSID of this handset.
PN: 1725-36038-001-H.doc
Troubleshooting
Message
Description
Incorrect Security settings.
Action
Verify that all the Security settings in the
handset match those in the APs.
No Net Found
xxxxxxxxxxxx yy
x…x = AP MAC address.
yy = AP signal strength.
Handset cannot find a suitable AP;
displays MAC and signal strength of
“best” non-suitable AP found.
Check AP and handset network settings
such as ESSID, Security, Reg domain
and Tx power.
Ensure APs are configured per
Configuration Guide.
Try Site Survey mode to determine a
more specific cause.
No Net Found
No CCX APs
The Wireless Telephone is configured
for CCX compatible operation, but
cannot find an access point that is
advertising CCX capability.
Check the AP configuration against
VIEW Configuration Guide for AP.
No Net Found
No CCKM APs
The Wireless Telephone is configured
to use CCKM for fast and secure
handoffs, but cannot find an access
point that is configured appropriately.
Check the AP configuration against
VIEW Configuration Guide for AP.
No Net Found
No WMM APs
The Wireless Telephone is configured
to use Wi-Fi Standard QoS, but cannot
find an AP configured appropriately.
Check the AP configuration against
VIEW Configuration Guide.
No PBX Response
The handset has exceeded its
retransmission limit with no ACK
response from proxy server.
Verify that proxy server IP address and
port are properly configured.
No Reg Domain
Regulatory Domain Not Set.
Configure the Regulatory Domain of the
handset.
No Server IP
In the case of static IP configuration,
the handset failed to find the call
server IP.
Verify that call server info is properly
configured on the handset.
No SIP TFTP IP
No IP address has been entered for
the SIP TFTP server.
In static IP mode the SIP TFTP server
address must be entered in the
administrative menus.
No SIP user file
The phone is attempting to download a
SIP configuration file from the SIP
TFTP server. A file must be available
for the username that was entered
either in the Admin menus or as
requested at power-on.
Ensure a SIP configuration file is
available on the SIP TFTP server and is
named as specified (sip_username.cfg).
No SVP IP
The handset is configured for “Static
IP” (as opposed to “use DHCP”), and
no valid SpectraLink 8000 SVP Server
address has been entered.
Enter a valid SpectraLink 8000 SVP
Server IP address in the configuration
setting or change to “use DHCP.”
PN: 1725-36038-001-H.doc
131
SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Message
No SVP Response
yyy.yyy.yyy.yyy
Description
y…y = SVP Server IP address.
Handset has lost contact with the SVP
Server.
Action
This may be caused by bad radio
reception or a problem with the
SpectraLink 8000 SVP Server. The
handset will keep trying to fix the
problem for 20 seconds, and the
message may clear by itself. If it does
not, the handset will restart. Report this
problem to the system administrator if it
keeps happening.
No SVP Server
Handset can’t locate SpectraLink 8000
SVP Server.
IP address configuration of SpectraLink
8000 SVP Server is wrong or missing.
SpectraLink 8000 SVP Server is not
working.
Check error status screen on
SpectraLink 8000 SVP Server.
No LAN connection at the SpectraLink
8000 SVP Server.
Verify SpectraLink 8000 SVP Server
connection to LAN.
No SVP Server
No DNS Entry
Handset unable to perform DNS
lookup for SVP Server, server had no
entry for SVP Server.
The network administrator must verify
that a proper IP address has been
entered for the SVP Server DHCP
option 151.
No SVP Server
No DNS IP
Handset unable to perform DNS
lookup for SVP Server, no IP address
for DNS server.
The network administrator must verify
proper DHCP server operation.
No SW Found
A required software component has
not been identified.
Check that the handset license type has
a corresponding entry in the
slnk_cfg.cfg file.
Check that the pd11sid.bin and
pi110000.bin entries exist in under this
license type in the slnk.cfg.cfg file.
No TFTP
Response
The handset could not get the TFTP
server to respond.
The handset will continue to boot
without checking if its current code is
the latest available. Check that the
TFTP server is operational. If the
Wireless Telephone is using DHCP,
check that the DHCP options are set
correctly.
No WPA
PassPhrase
This error only appears when the
Admin Menus are exited. The handset
is configured for WPA-PSK or WPA2PSK and no pass phrase or shared
key has been entered.
Enter the pass phrase or pre-shared
key and restart the handset
Not Installed!
A required software component is
missing.
Check that all required software files
are on the TFTP server, if over-the-air
downloading is being used. If the error
repeats, contact Polycom Technical
Support.
Press END
The far end of a call has hung up.
Hang up the near end.
132
PN: 1725-36038-001-H.doc
Troubleshooting
Message
Press END to quit
Description
The handset is waiting to acquire
bandwidth required for voice
communication.
Action
Press END or wait until bandwidth is
available.
Prom Bad Length
The handset software downloaded files
that are incorrect or corrupted.
Download new software from the OEM
site per Software Maintenance.
Registering
The handset has completed
initialization of the SIP application and
is attempting to register lines to the
SIP proxy servers.
If registrations are failing, the phone can
stay in this state for a considerable
length of time. After the phone leaves
this state, press the LINE key to view
what lines have failed to register.
Ensure usernames and passwords
have been entered in administrative
menus for registrations that have failed
and that proxy information is correct in
the SIP configuration files.
RTP Open Failed
The handset attempted to open an
RTP port for audio but was
unsuccessful.
Verify that SpectraLink 8000 SVP
Server capacity has not been
exceeded.
Select License
The correct protocol has not been
selected from the license set.
Using the Admin menu, select one
license from the set to allow the phone
to download the appropriate software.
Server Busy
Handset is attempting to download
from a TFTP server that is busy
downloading other devices and
refusing additional downloads.
None, the handset will automatically
retry the download every few seconds.
SIP Login
Prompt for login information –
username and password.
At power-on initialization, no username
was detected in the Admin menu items
for SIP registrations. Enter a valid
username and password for an existing
SIP configuration file.
Skt Open Fail
Socket open fail. Occurs when the
handset attempts to open a connection
to the proxy server but fails.
Verify that SpectraLink 8000 SVP
Server capacity has not been
exceeded.
Service Rej.
The SpectraLink 8000 SVP Server has
rejected a request from the handset.
The handset will restart and attempt to
re-register with the SpectraLink 8000
SVP Server, which should fix the
problem. Report to your administrator if
it keeps happening.
Storing Config
Handset is storing changes to handset
configuration.
None. Informational only. The handset
may display this briefly following a
configuration change or software
download.
SVP Service Rej.
The SpectraLink 8000 SVP Server has
rejected a request from the handset.
The handset will restart and attempt to
re-register with the SVP Server, which
should fix the problem. Report to your
administrator if it keeps happening.
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133
SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Message
System Busy
yyy.yyy.yyy.yyy
Description
y…y = SVP Server IP Address.
SVP Server has reached call capacity.
Action
All call paths are in use, try the call
again in a few minutes.
System Locked
(with Busy Tone)
SpectraLink 8000 SVP Server is
locked.
Try call again later, system has been
locked for maintenance.
TFTP
ERROR(x):yy
A failure has occurred during a TFTP
software download. (x) = The file
number which was being downloaded;
yy is an error code describing the
particular failure. Possible error codes
are:
Error code 01, 02, 07, or 16 - check the
TFTP server configuration.
01 = TFTP server did not find the
requested file.
02 = Access violation (reported from
TFTP server).
Error code 81, the handset will attempt
to download the file again.
For other messages, power off the
handset, then turn it on again to retry
the download. If the error repeats, note
it and contact Polycom Customer
Support.
07 = TFTP server reported "No such
user" error.
16 = No TFTP server address.
81 = File put into memory did not CRC.
FF = Timeout error. TFTP server did
not respond within a specified period of
time.
Too Many Errors
The handset continues to reset and
cannot be recovered.
Fatal error. Return handset to Polycom.
Unknown
xx:yy:zz
A phrase is missing from the phintl file.
Download new software from the
Polycom website per Software
Maintenance.
Updating …
The handset is internally updating its
software images.
None. The handset may do this briefly
after a download. This is informational
only.
Updating Code…
Handset is downloading new software
into memory. The number icons at the
bottom of the display indicate which file
number is currently being downloaded.
This message also displays a progress
bar. When the progress bar fills the
display line the update operation is
complete on that file.
None. When the progress bar fills the
display line the update operation is
complete on that file.
Wait for bandwidth
The phone is waiting for bandwidth
sufficient for voice communication.
No action required. You will have the
option of pressing END to abort the
phone call.
Waiting…
Handset has attempted some
operation several times and failed.
None. The handset is waiting for a
specified period of time before
attempting that operation again.
Wrong Code Type
The software loaded into the handset
is incorrect for this model phone.
Replace the software image on the
TFTP server with software that is
correct for the handset model.
134
Do not turn the handset off during this
operation.
PN: 1725-36038-001-H.doc
Appendix A: Regulatory Domains
This table details the specifications for regulatory domain settings.
Polycom recommends that you check with local authorities for the
latest status of their national regulations for both 2.4 and 5 GHz
wireless LANs.
Domain
Identifier
01
100mW (+20dBm)
5.1500 – 5.2500 GHz
36 – 48
No
50mW (+17dBm)
5.2500 – 5.3500 GHz
52 – 64
Yes
5.4700 – 5.7250 GHz
100 – 140
Yes
5.7250 – 5.8250 GHz
149 – 161
No
1 – 13
n/a
5.1500 – 5.2500 GHz
36 – 48
No
5.2500 – 5.3500 GHz
52 – 64
Yes
5.4700 – 5.7250 GHz
100 – 140
Yes
1 – 13
n/a
5.1500 – 5.2500 GHz
36 – 48
No
5.2500 – 5.3500 GHz
52 – 64
Yes
1 – 13
n/a
5.1500 – 5.2500 GHz
36 – 48
No
5.2500 – 5.3500 GHz
52 – 64
Yes
1 – 13
n/a
5.1500 – 5.2500 GHz
36 – 48
No
5.2500 – 5.3500 GHz
52 – 64
Yes
5.4700 – 5.6500 GHz
100 – 124
Yes
5.7250 – 5.8250 GHz
149 – 161
No
g only
b & b/g mixed
a
05
n/a
g only
b & b/g mixed
a
04
1 – 11
g only
b & b/g mixed
a
03
Max. Power Limit
(peak power)
Band
g only
b & b/g mixed
a
02
DFS
Required?
802.11 Mode
g only
b & b/g mixed
a
PN: 1725-36038-001-H.doc
Channels
Countries
US
Canada
Brazil
100mW (+20dBm)
Europe
Australia
100mW (+20dBm)
New Zealand
Japan
100mW (+20dBm)
Singapore
100mW (+20dBm)
Korea
100mW (+20dBm)
135
SpectraLink 8020/8030 Wireless Telephone: Administration Guide for SIP
Domain
Identifier
802.11 Mode
Band
g only
b & b/g mixed
06
a
a
a
Max. Power Limit
(peak power)
Countries
Taiwan
n/a
5.2500 – 5.3500 GHz
52 – 64
Yes
5.4700 – 5.7250 GHz
100 – 140
Yes
5.7250 – 5.8500 GHz
149 – 165†
No
1 – 13
n/a
100mW (+20dBm)
5.1500 – 5.2500 GHz
36 – 48
No
50mW (+17dBm)
5.2500 – 5.3500 GHz
52 – 64
Yes
5.4700 – 5.7250 GHz
100 – 140
Yes
5.7250 – 5.8250 GHz
149 – 161
No
1 – 11
n/a
5.1500 – 5.2500 GHz
36 – 48
No
5.2500 – 5.3500 GHz
52 – 64
Yes
g only
b & b/g mixed
08
DFS
Required?
1 – 11
g only
b & b/g mixed
07
Channels
100mW (+20dBm)
Hong Kong
100mW (+20dBm)
Mexico
India
100mW (+20dBm)
5.7250 – 5.8500 GHz
149 – 161
No
† Channel 165 is not currently supported on the handset when the UNI-3 (5.7250 – 5.8250) band is enabled for 802.11a.
136
PN: 1725-36038-001-H.doc
Index
A
Access point
Coverage .........................................................106
Coverage test ..................................................117
Access point, description .....................................15
C
Capacity ...............................................................123
Checking code .....................................................122
Customer Support Hotline ....................................4
OAI Server IP..............................................34, 38
Obstructions ........................................................124
Out of range.........................................................123
R
Regulatory Domain ..............................................41
Restore Defaults ..............................................44, 45
S
Site Certification..................................................117
Status messages...................................................125
D
T
Download messages
Failure or Recover..........................................122
Normal ............................................................122
Transmission obstructions.................................124
E
Erasing memory..................................................122
Ethernet Switch, description ...............................16
H
Hotline......................................................................4
U
Updating code.....................................................122
W
Wireless Telephone
Problems with ................................................123
Status messages..............................................125
WT, description.....................................................15
O
OAI
OAI Gateway....................................................34
OAI On/Off ......................................................34
PN: 1725-36038-001-H.doc
137